| 00:01.05 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.24) |
| 00:02.31 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 00:02.37 | *** join/#asterisk sariabod (~sariabod@ip21.farheap.net) |
| 00:02.54 | Ayano | does anyone know of any irc rooms that would know about xpl? |
| 00:04.13 | ManxPower | stustu, It's device dependent |
| 00:04.24 | ManxPower | stustu, And mostly on cheap/free phones |
| 00:04.40 | ManxPower | ~docs |
| 00:04.41 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 00:06.16 | *** join/#asterisk anthm (~anthm@70.8.125.199) |
| 00:06.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:06.41 | hardwire | blah |
| 00:06.54 | hardwire | I should try xchat-gnome |
| 00:09.20 | sariabod | anybody know the usual times I can catch markster online? |
| 00:10.49 | *** join/#asterisk xstaci (xstaci@216.199.122.213) |
| 00:10.53 | hardwire | just stay up for the next 72 hours and see if you can find him. |
| 00:11.16 | sariabod | oh good, and I thought it was going to be difficult |
| 00:11.29 | stustu | How do I set up three way calling? Do I need to use MeetMe, or is there a simpler way to connect two people that I've dialled? |
| 00:12.03 | xstaci | hi peoples, wondering if anyone uses APM+* and could help me with a problem |
| 00:12.03 | hardwire | the client can typically handle that |
| 00:12.57 | *** join/#asterisk jekil (~jekil@host152-172.pool8252.interbusiness.it) |
| 00:13.21 | *** join/#asterisk dcb1 (~dcb@CPE-60-231-180-246.vic.bigpond.net.au) |
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| 00:16.22 | Darwin35 | night guys off to my sleep study group . boy this will be fun |
| 00:16.31 | file | ...hi |
| 00:16.42 | BoRiS | file!!!!!!!!!!!!! Darwin35!!!!!!! |
| 00:16.50 | stustu | hardwire: Well, not my analog phones at home... :) |
| 00:17.02 | Darwin35 | Boris |
| 00:17.05 | xstaci | my dialparties.agi does not run, just kinda sits there and does nothing, no output at all |
| 00:17.07 | Darwin35 | chat tomarrow |
| 00:17.07 | hardwire | stustu: there are ways |
| 00:17.08 | BoRiS | Wasssssssssssup? |
| 00:17.09 | hardwire | it is written |
| 00:17.15 | Darwin35 | sleep study night |
| 00:17.20 | hardwire | I think voip-info.org is your answer |
| 00:17.27 | hardwire | as well as learning the digium * codes |
| 00:17.30 | Darwin35 | night at the hospital why they monitor my sleep |
| 00:17.35 | hardwire | *69, etc.. |
| 00:17.54 | hardwire | but I would say.. if you wanna do three way calling |
| 00:17.54 | Darwin35 | last night of it |
| 00:17.57 | hardwire | have them dial your meetme |
| 00:19.26 | stustu | hardwire, I realize the equivalent to my phone handling it might be the zapata module handling it for them... |
| 00:20.11 | *** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar) |
| 00:20.56 | BoRiS | :-))) |
| 00:21.50 | eipi | my mobo pci 2.1, has detected my tdm board... |
| 00:22.04 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 00:22.28 | hardwire | file: whats happeneeeeen |
| 00:23.02 | file | oh, not much |
| 00:23.11 | hardwire | I enabled per nick colors |
| 00:23.13 | hardwire | your a pretty one |
| 00:23.16 | file | lol |
| 00:23.25 | hardwire | very girly |
| 00:24.17 | *** join/#asterisk BigCanOfTuna (~chatzilla@dsl-macn-66-18-205-30-cgy.nucleus.com) |
| 00:24.39 | opus___ | Has anyone had this problem with AMP: http log |
| 00:24.43 | opus___ | <PROTECTED> |
| 00:24.45 | BoRiS | file: Already at the hotel? |
| 00:25.13 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
| 00:25.14 | xstaci | looking for help getting dialparties.agi to run correctly |
| 00:25.31 | JunK-C | xstaci: which language ? perl ? |
| 00:25.38 | BigCanOfTuna | I am using the manager API to automatically call my work number, however, as soon as my phone rings the extension proceeds to execute, is there a way that I can only begin executing the extension when I actually pick up the phone? |
| 00:25.46 | xstaci | believe so, when i run it it sits there with no output |
| 00:26.09 | JunK-C | xstaci: believe? if you dont know if its an perl agi, ya've trouble. |
| 00:26.13 | xstaci | it is perl |
| 00:26.25 | xstaci | i can't run dialparties.agi or agi-test.agi or any other agi files |
| 00:26.36 | xstaci | other .pl scripts seem to run though |
| 00:26.40 | xstaci | just nothing agi related |
| 00:26.41 | JunK-C | perl -c dialparties.agi gives ? |
| 00:26.50 | SPoon_TSX | hi everyone. just wondering if someone can tell how many phone call I can make at the same time on a single span T1 connection? |
| 00:27.02 | xantus | 24 or 23 |
| 00:27.06 | xstaci | syntax OK |
| 00:27.28 | JunK-C | and ls -l dialparties.agi? |
| 00:27.40 | JunK-C | SPoon_TSX: if its a PRI, 23, if its a t1, 24 |
| 00:27.46 | xstaci | -rwxr--r-- 1 asterisk asterisk 7683 Mar 10 14:00 dialparties.agi |
| 00:28.04 | SPoon_TSX | I am so new to asterisk, please forgive my stipud questions. |
| 00:28.05 | xantus | 755 it |
| 00:28.26 | SPoon_TSX | Seems like PRI is more expensive then T1, is it true? |
| 00:28.26 | JunK-C | SPoon_TSX: voip-info.org will be a good start. |
| 00:28.33 | JunK-C | no |
| 00:28.39 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 00:28.39 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 00:28.44 | SPoon_TSX | PRI is cheaper? |
| 00:28.46 | xantus | SPoon_TSX: sometimes, cause you get CID on PRI |
| 00:28.54 | SPoon_TSX | oic. |
| 00:29.00 | JunK-C | xstaci: whats ur cli's output? |
| 00:29.13 | SPoon_TSX | Roughly how much would it cost me on a PRI line? Any idea? |
| 00:29.16 | xstaci | here, http://sourceforge.net/forum/forum.php?thread_id=1246195&forum_id=414452 |
| 00:29.21 | xstaci | most of it is explained there |
| 00:29.31 | xantus | i was able to set my outgoing CID on a PRI |
| 00:29.34 | JunK-C | SPoon_TSX: ask ur telco! |
| 00:29.36 | xstaci | just 755'ed dialparties.agi and it still just sits there, doesnt exit or give any output |
| 00:30.05 | xantus | whats your dialparties.agi look like? can you nopaste it? |
| 00:30.06 | SPoon_TSX | My telco company in my local area know NOTHING. They even told me they never hear about VoIP. Damn! |
| 00:30.21 | xantus | http://nopaste.snit.ch:8001/ |
| 00:30.21 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 00:30.25 | JunK-C | SPoon_TSX: isnt about voip. |
| 00:30.32 | JunK-C | its about normal telecom, no? |
| 00:30.42 | mandreko | yup |
| 00:30.51 | JunK-C | then then should know it. |
| 00:30.52 | file | oh no, VON is over |
| 00:30.55 | mandreko | yes |
| 00:30.55 | xstaci | xantus: what channel should i choose for nopaste? |
| 00:30.59 | *** join/#asterisk tuxinator_linuxM (~anonymous@m190e36d0.tmodns.net) |
| 00:31.02 | xantus | they never heard of * when i was putting in a PRI a year ago |
| 00:31.04 | JunK-C | file: any pics somewhere? |
| 00:31.08 | file | http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view |
| 00:31.12 | SPoon_TSX | Yes, I call them to see they offer a complete package for IP phone system and they told me they never heard we can make a phone call over Internet. |
| 00:31.13 | xantus | #poe is fine |
| 00:31.14 | file | that's for the Digium booth |
| 00:31.31 | tuxinator_linuxM | hey file, long time no see |
| 00:31.35 | file | hey hey |
| 00:31.43 | file | tuxinator_linuxM: we're in the Hyatt, room 503 |
| 00:31.48 | tuxinator_linuxM | On my way |
| 00:31.52 | mandreko | SPoon_TSX: just tell them you want a PRI, which is just like a bundle of 23 phone lines |
| 00:32.01 | JunK-C | SPoon_TSX: ya should care about it. take a PRI. |
| 00:32.19 | *** join/#asterisk Hydr0p0nX (~hydr@482-brhm1.adsl.wwisp.net) |
| 00:32.37 | *** join/#asterisk pulu (~chatzilla@65.77.78.3) |
| 00:32.38 | Hydr0p0nX | sup |
| 00:33.06 | hardwire | pulu is a food isn't it? |
| 00:33.07 | Hydr0p0nX | anyone know of a good app for monitoring how many pri channels are in use ? |
| 00:33.20 | hardwire | cli |
| 00:33.27 | potter | asterisk -rv |
| 00:33.31 | JunK-C | file: we're means? |
| 00:33.49 | file | JunK-C: we are |
| 00:33.56 | file | twisted, drumkilla, and I |
| 00:33.58 | xstaci | xantus, get my msg? |
| 00:34.17 | JunK-C | say hi to them. |
| 00:34.32 | SPoon_TSX | Sweet. |
| 00:34.54 | SPoon_TSX | Anyone have any sucess on PBX 2 office total 20 extension using Asterisk and Digiam hardware? |
| 00:34.58 | eipi | question: i received my tdm400p with 2 fxo on tel 3 and tel 4... in zaptel.conf what i have to add? i tried fxoks=3-4 but i receive this on trying to start zaptel: ZT_CHANCONFIG failed on channel 3: Invalid argument |
| 00:35.07 | xantus | yes |
| 00:35.40 | JunK-C | eipi: go read about configuration, FXO uses FXS signalling. |
| 00:35.51 | SPoon_TSX | JunK-C: Which card you will suggest me to get for my new office which will have a 1 PRI and 20 extensions? |
| 00:36.08 | pulu | pulu can mean beef, so you could say it's a food |
| 00:36.16 | eipi | junk-c ok |
| 00:36.19 | pulu | but please do not attempt to eat me, i only leave those rights for the wife. ;-) |
| 00:37.22 | JunK-C | T100P |
| 00:37.23 | SimonR | anyone know a good voip termination provider that will handle a large number of simultaneous calls? |
| 00:37.46 | pulu | i've never done a whole lot of concurrent calls but it seems like that's the main kind of business that voipjet is looking for |
| 00:38.14 | SPoon_TSX | Do you mean this one? http://www.digium.com/index.php?menu=wildcard_te110p |
| 00:39.24 | drumkilla | file!!!!!!!!!!! |
| 00:39.32 | file | drumkilla!!!!!!!!!!! |
| 00:39.42 | SimonR | pulu, yes, voipjet mentioned that they have some customers doing large numbers of calls. |
| 00:39.58 | SimonR | they said they have a couple of providers who can handle big spikes in traffic. |
| 00:40.16 | SimonR | do you know any others? |
| 00:40.35 | SimonR | I'm assuming people that use softswitch-based networks are the best bet, like a Level3 reseller. |
| 00:43.17 | file | eek ugh KABOOM |
| 00:43.24 | SPoon_TSX | Quick question is Digi. still selling T100P? |
| 00:44.16 | JunK-C | its renamed TE101P i think |
| 00:45.06 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 00:46.56 | SPoon_TSX | Thank you so much, JunK-C you are SO Helpful!! |
| 00:47.01 | ctooley | SimonR, TXLink has done a good job for large spikes for us. |
| 00:47.23 | Hydr0p0nX | anyone know of a good article on fxo/fxs differences ? |
| 00:48.37 | SPoon_TSX | I am about to make purchase for TE110P. Just want to make sure that I can use this Card to serve my PRI line which I can archive 23 concurrent call? |
| 00:48.42 | JunK-C | SPoon_TSX: no problem houston. |
| 00:48.46 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:49.04 | JunK-C | right. |
| 00:49.14 | SPoon_TSX | JunK-C: Thanks! |
| 00:49.34 | JunK-C | and if u plans to have many PRI, ya can take a look on 4spans T1s. |
| 00:49.43 | JunK-C | (TE405P if u care) |
| 00:49.49 | Hydr0p0nX | or can someone break it down for me .... i was selected to handle all the voip config/install/engineering for customers and only know how to punch down as far as telco goes heh |
| 00:51.08 | SPoon_TSX | cool. |
| 00:51.27 | SPoon_TSX | Regarding the PRI line, I just need a standard ISDN PRI line? |
| 00:53.42 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.248) |
| 00:54.31 | *** part/#asterisk search_learn2005 (~Miranda@209.68.139.150) |
| 00:56.59 | *** part/#asterisk dcb1 (~dcb@CPE-60-231-180-246.vic.bigpond.net.au) |
| 00:58.25 | xstaci | having trouble getting dialparties.agi, can anyone help? |
| 00:58.45 | JunK-C | xstaci: Ive asked ya for CLI's output before. |
| 00:59.05 | xstaci | junk-c: http://sourceforge.net/forum/forum.php?thread_id=1246195&forum_id=414452 |
| 00:59.16 | xstaci | thought i pasted that before |
| 00:59.53 | xstaci | thats all that shows in the cli when i call * |
| 01:01.05 | *** join/#asterisk Milligan (~noc@wkstn2.gnwd-noc.valuelinx.net) |
| 01:03.25 | Milligan | Is there anyone here that is interested in me hiring them to assist me in getting our Asterisk servers in operation? |
| 01:03.25 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 01:03.51 | paulc | Milligan: I might be - depending on the set up etc |
| 01:04.42 | Milligan | I have quite a bit of it setup, but I am needing to get the call queues worked out.... I have not had time to learn alot about them |
| 01:04.55 | Milligan | I have the extensions and the voicemail setup but that is pretty much it. |
| 01:05.25 | SexyKen | Hey guys. |
| 01:05.31 | SexyKen | Anyone got much experience with CDR? |
| 01:05.34 | *** join/#asterisk grumpie (~vijay@dsl093-139-122.sfo4.dsl.speakeasy.net) |
| 01:05.43 | *** join/#asterisk JmanA9 (~josh@h60.178.40.69.ip.alltel.net) |
| 01:05.53 | Hydr0p0nX | anyone know much about audiocodes mp series IAD's ? |
| 01:06.30 | grumpie | Hi folks ... has anyone seen oej around on this channel? |
| 01:06.58 | grumpie | Hyd ... sorry ...I'm not even sure I understand your qn. :-( |
| 01:07.18 | Damin_Mobile | No, but I saw him an hour ago.. |
| 01:07.25 | Damin_Mobile | at von... |
| 01:07.34 | Nugget | is "question" really that much of a burden to type? |
| 01:07.59 | Nugget | I can totally understand not wanting to type "Hydr0p0nX" |
| 01:08.01 | SexyKen | I am having some trouble figuring out how the CDR stores data...and how to link calls together. Each call seems to have multiple db entries, how would I link entries to the same call? |
| 01:08.57 | *** join/#asterisk adeilt (wil@pcp04615196pcs.gambrl01.md.comcast.net) |
| 01:09.09 | Hydr0p0nX | just wondering if they are decent iad's really, we resell for another company and they recommended these iad's as possibilities but after looking at the price i'm starting to think we'd do better with an adit box to handle analog phones if need be |
| 01:09.24 | file | Damin_Mobile: poke |
| 01:10.07 | file | I think twisted and tuxinator_linuxM are falling asleep |
| 01:10.51 | file | oh tuxinator_linuxM recovered |
| 01:10.58 | file | ah and twisted |
| 01:12.10 | Milligan | What causes the line to ring when you hit the messages button on the phone instead of going directly to voicemail....? |
| 01:13.46 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 01:15.56 | Milligan | anyone have any ideas where that might be.... |
| 01:16.41 | *** join/#asterisk zignig (~simon@203.217.15.10) |
| 01:16.52 | Godsey | I was having a problem w/ building zaptel from cvshead this morning |
| 01:16.58 | *** join/#asterisk boch (ROK@host115.200.61.130.ifxnw.com.ar) |
| 01:17.00 | Godsey | turned out to be a problem w/ distcc and ccacache :) |
| 01:17.20 | SexyKen | I am having some trouble figuring out how the CDR stores data...and how to link calls together. Each call seems to have multiple db entries, how would I link entries to the same call? |
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| 01:21.44 | Godsey | might anyone be able to reach iaxtalk.com's website? |
| 01:22.04 | *** join/#asterisk IQ (~IQ@70-59-160-90.omah.qwest.net) |
| 01:22.21 | KalD|Work | OMG... the CTO just brought me * mouse pads from VON! HAHAHA He loves * ... this is good - no more win32 bs for us! |
| 01:22.58 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 01:23.17 | potter | how do you catch a no answer using the dialplan priority? anyone? |
| 01:24.55 | grumpie | damin_mobile ... thanks for the tip. i'm glad he is still here. |
| 01:25.27 | adeilt | any recommendation for a beginner HOWTO? My job involves satellite communications, and I had a brief course a couple of years ago on phone switching, but i'm having a little trouble getting off the ground with asterisk... |
| 01:25.44 | adeilt | i'm just looking to do voip with software phones to start with... just playing around |
| 01:27.27 | *** join/#asterisk |Vulture| (Vulture@dhcp64-134-129-79.wbt.phx.wayport.net) |
| 01:27.57 | SexyKen | I am having some trouble figuring out how the CDR stores data...and how to link calls together. Each call seems to have multiple db entries, how would I link entries to the same call? |
| 01:28.36 | mishehu | SexyKen: my CDR's only have one row per call. |
| 01:28.50 | |Vulture| | hey guys I am trying to order a PRI and I asked them what signaling type they used and I got back the following: http://pastebin.ca/7210 |
| 01:28.53 | |Vulture| | does that sound right? |
| 01:29.07 | jontow | adeilt; whats the basic trouble you're having? |
| 01:29.08 | mishehu | when you say multiple db entries, do you mean database rows > 1 or do you mean it appears in more than one column on the same row |
| 01:29.39 | adeilt | just wondering where to start with the available docs |
| 01:30.04 | adeilt | i know that i need to do the reading, but i'm having trouble finding the "Start Here, Fool" page |
| 01:30.33 | SexyKen | mishehu -> How did you set that up? |
| 01:31.02 | SexyKen | mishehu -> Well, I'm just kind of confused. |
| 01:31.26 | SexyKen | mishehu -> I wish someone could explain exactly what the columns in the cdr database are used for. |
| 01:31.27 | *** join/#asterisk purplebob (~don@207-40-115-38.sugardog.com) |
| 01:31.37 | |Vulture| | anyone here take a look at my PRI question? |
| 01:31.55 | purplebob | having a blast with my new snom 190's here at the office. |
| 01:32.27 | purplebob | having one issue with getting the programable buttons to display another extension status. |
| 01:32.32 | SexyKen | I dont see anyplace to configure cdr either. |
| 01:32.57 | purplebob | getitng failed to authenticate user blah@blash for SUBSCRIBE |
| 01:33.23 | jontow | adeilt; the best way to start is to download a copy of the sources, attempt to compile and install it on an unused machine and then install the sample config files and start reading through them.. they're very well commented |
| 01:33.29 | shmaltz | SexyKen, I dont' think you can configure CDR in any way. CDR needs lots of fixing up |
| 01:33.38 | jontow | adeilt; keep in mind, while you're browsing the config files, you should cross-reference them with http://www.voip-info.org/ |
| 01:33.40 | jontow | @ docs |
| 01:33.43 | jontow | ~ doc |
| 01:33.44 | jbot | from memory, doc is The command is "~docs", moron! |
| 01:33.48 | jontow | ~docs |
| 01:33.49 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 01:33.51 | jontow | (jbot)++ |
| 01:34.05 | jontow | go with some of those.. i remember having good luck with handbook-draft.pdf as reading material. |
| 01:34.27 | SexyKen | shmaltz -> Do you have cdr working in a way that you like? |
| 01:34.29 | adeilt | debian made the install easy, but is there a significant advantage to building from source? |
| 01:34.37 | shmaltz | nope |
| 01:34.40 | shmaltz | look at this: |
| 01:34.43 | purplebob | I followed the wiki example. makes sense.... rebooted the phone.... if I push the botton the monitored ext rings... just no status and getting the error. |
| 01:34.49 | jontow | yeah.. if you get the latest cvs copy.. you get the latest bugfixes and features |
| 01:34.59 | adeilt | k... i'll give that a try |
| 01:34.59 | jontow | prebuilt packages are most often NOT up to date.. |
| 01:35.04 | shmaltz | http://lists.digium.com/pipermail/asterisk-users/2005-February/092049.html |
| 01:35.24 | jontow | i know the freebsd one isn't.. i'd bet the debian one isn't :) |
| 01:35.37 | adeilt | true... they serve for a lot of things, but now that i think about it, featureful telephone switches tend to attract an... unsavory crowd |
| 01:35.42 | file | h;kh;k |
| 01:35.45 | sariabod | exit |
| 01:35.47 | sariabod | exit |
| 01:35.47 | shmaltz | SexyKen, I think I will have to enable the user field and work from there. I don't think it will fix my problem |
| 01:35.50 | sariabod | exit |
| 01:35.52 | sariabod | quit |
| 01:35.58 | adeilt | updated security patches would be a good thing |
| 01:36.20 | shmaltz | SexyKen, keep in mind that SIP has it's own set of problems in billing |
| 01:36.40 | purplebob | any thoughts? |
| 01:36.59 | SexyKen | shmaltz -> Well I'm trying to figure out a way to explain the problem better. |
| 01:37.55 | purplebob | chan_sip.c:7574 handle_request: Failed to authenticate user <sip:2001@192.168.2.254>;tag=syoh6uuj9u for SUBSCRIBE |
| 01:39.05 | *** join/#asterisk jcwunder (~chatzilla@diale141.ppp.lrz-muenchen.de) |
| 01:39.10 | shmaltz | SexyKen, you only see multiple rows, if/when you transfer one leg of a call to an extension in the dialplan that does this, like conferencing. But when you transfer to another extension, it should be all in one row, although I wish it would be in more than one, like I explain in my post |
| 01:39.54 | jcwunder | who is using the recent debian zaptel-source with recent bristuff's zaphfc ? |
| 01:40.04 | SexyKen | shmaltz -> I am working with queues, and it seems to be all messed up. |
| 01:40.56 | shmaltz | looks like queues might have the same problem, the way I troubleshot it was to rename Master.cvs and have * record from fresh, and then I started playing around. |
| 01:41.31 | shmaltz | SexyKen, please work on this, together we might have better chances on getting this fixed. |
| 01:42.18 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 01:44.22 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 01:48.27 | *** join/#asterisk FuriousGeorge (~Brian@ool-43516ebb.dyn.optonline.net) |
| 01:50.24 | SexyKen | CDR sucks. |
| 01:51.10 | FuriousGeorge | can anyone point me to a resource a little more general than the asterisk handbook draft sos i can learn me some of this magic |
| 01:51.41 | hardwire | I bought a book FuriousGeorge |
| 01:51.58 | FuriousGeorge | hardwire: yeah, which one? |
| 01:52.03 | hardwire | the good book. |
| 01:52.16 | FuriousGeorge | i didnt know asterisk is spoken for in the bibge! |
| 01:52.21 | hardwire | indeed |
| 01:52.23 | hardwire | its a faith |
| 01:52.28 | FuriousGeorge | and therea re people who say it shouldnt be interpreted literally |
| 01:52.32 | hardwire | god actualy smote people who smoke it |
| 01:52.46 | hardwire | http://b2b.abptech.com/images/abp/asteriskbook_h.jpg |
| 01:53.45 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 01:53.55 | FuriousGeorge | hardwire, do you think that might be advanced. i can barely use linux |
| 01:53.56 | *** join/#asterisk cced (~wangxinta@222.33.36.198) |
| 01:54.14 | FuriousGeorge | makes me wanna fstab myself, and fomit all over my frame pointer |
| 01:54.14 | cced | hi wildcard0 |
| 01:55.22 | jontow | i gawk at the hier(7) notion. |
| 01:56.02 | jontow | that book is not bad |
| 01:56.05 | jontow | i have a copy in my desk here.. |
| 01:56.11 | jontow | quite a few typos, but good info all around |
| 01:56.19 | jontow | its an introduction, not an in-depth coverage of everything * |
| 01:56.31 | jontow | helped me along the way repeatedly.. i recommend :) |
| 01:57.33 | SexyKen | Why dst is CDR ever be 's' |
| 01:58.13 | JunK-C | SexyKen: its the standard dst |
| 01:58.54 | *** join/#asterisk viLeR (1000@ip-47-233.telesat.com.co) |
| 01:58.55 | SexyKen | Junk-C -> What exactly does that mean? Some show an actual number or extension. Some just show 's'. |
| 01:58.59 | hardwire | Funbags: if you can barely use linux.. get a linux book |
| 01:59.27 | |Vulture| | SexyKen: if they are sent to that extension directly it shows that, if they ring into s, then they show s |
| 02:00.06 | |Vulture| | SexyKen: you can override it if you want, say if it is sent to s, then 102 answers it, it will say s unless transfered to s |
| 02:00.13 | FuriousGeorge | what to typicall business pbx's run over? cat3? |
| 02:00.23 | BoRiS | cat5 |
| 02:00.30 | hardwire | not relaly |
| 02:00.33 | BoRiS | cat3.......ugh |
| 02:00.35 | hardwire | atleast where I have seen |
| 02:00.42 | |Vulture| | no cat3 usually |
| 02:01.02 | SexyKen | Vulture -> Can you help me get my CDR setup properly because I'm extremelly confused. |
| 02:01.34 | |Vulture| | SexyKen: a lil help I am out of town right now |
| 02:01.56 | SexyKen | Mind if I send you an informative message so you can take a look at the currents? |
| 02:02.52 | |Vulture| | ya |
| 02:02.55 | |Vulture| | you can |
| 02:03.32 | *** join/#asterisk xeet2 (~xeet3@gw1.istx.net) |
| 02:05.14 | *** part/#asterisk Milligan (~noc@wkstn2.gnwd-noc.valuelinx.net) |
| 02:05.31 | *** join/#asterisk d00gster (~in_ter@67.71.121.232) |
| 02:06.30 | SPoon_TSX | hi everyone. Any recommendation on Hardware SIP Phone for business? |
| 02:06.32 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 02:07.11 | *** join/#asterisk JmanA9 (~josh@h60.178.40.69.ip.alltel.net) |
| 02:07.35 | xeet2 | spoon: cisco 7960 |
| 02:07.37 | *** part/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
| 02:08.36 | SPoon_TSX | xeet2: It is about $400 each, a little bit expensive. any other choose? |
| 02:08.51 | xeet2 | spoon: you're looking at the wrong place then, we get them for 260 |
| 02:09.11 | JunK-C | SPoon_TSX: Polycom SIP 500. |
| 02:09.15 | SPoon_TSX | USD? 260? Where? It Cheap! |
| 02:09.40 | xeet2 | spoon: yes, usd, hang on |
| 02:10.08 | SPoon_TSX | JunK-C: Rougly how much will it be? |
| 02:10.43 | JunK-C | SPoon_TSX: if i remember ok, i've paid like 280$CND |
| 02:10.49 | shmaltz | SPoon_TSX, voipsupply.com |
| 02:10.50 | *** join/#asterisk Dawson64 (Dawson64@24.247.102.233.gha.mi.chartermi.net) |
| 02:10.54 | JunK-C | 200$US maybe. |
| 02:11.14 | xeet2 | a bunch of places have 7960s for good prices |
| 02:11.34 | shmaltz | http://www.voipsupply.com/product_info.php?cPath=95_107&products_id=54 |
| 02:11.37 | shmaltz | $259 |
| 02:12.05 | xeet2 | is there any way to/application that can determine how many current active calls exist for a given number, on a pri? |
| 02:12.15 | JunK-C | xeet2: yes |
| 02:12.18 | JunK-C | i did a patch for that |
| 02:12.21 | JunK-C | see the bug tracker |
| 02:12.26 | FuriousGeorge | are these fxo/fxs modules chips that go on the tdm400p? |
| 02:12.29 | JunK-C | not yet implemented in HEAD. |
| 02:12.36 | JmanA9 | when i'm trying to dial out to fwd using the instructions on the getting started guide, i get this message in the asterisk terminal, and a 403 forbidden error in my softphone: |
| 02:12.36 | JmanA9 | <PROTECTED> |
| 02:12.36 | JmanA9 | Mar 10 21:11:47 WARNING[6837]: pbx.c:1554 pbx_extension_helper: No application 'Dial(IAX2/${FWDNUMBER}' for extension (sip, 393611, 2) |
| 02:12.36 | JmanA9 | <PROTECTED> |
| 02:12.45 | JmanA9 | anyone know what would cause that? |
| 02:12.51 | shmaltz | http://www.voipsupply.com/product_info.php?cPath=95_109&products_id=356 |
| 02:12.53 | shmaltz | $299 |
| 02:13.00 | xeet2 | junk: is it an application or like a global variable? |
| 02:13.06 | JmanA9 | happens with any fwd number |
| 02:13.34 | JunK-C | xeet2: http://bugs.digium.com/bug_view_page.php?bug_id=0003679 |
| 02:13.36 | xeet2 | jmana9: you're missing a ) at the end of your dial command |
| 02:13.45 | SPoon_TSX | comparing Cisco vs PolyCOM, which one have better sound quality? |
| 02:13.47 | JmanA9 | ok, i'll check that out, thanks |
| 02:13.51 | JunK-C | ya can do show extension <did> |
| 02:13.58 | JunK-C | to know how many calls are in that did |
| 02:15.35 | FuriousGeorge | can someone explain to me what the difference between a pci fxo/fxs, and a fxo/fxs module? is the latter just a chip we place on the tdm400p? |
| 02:15.48 | mgth | yes |
| 02:16.12 | FuriousGeorge | and all the ports on that device become fxos or fxss |
| 02:16.13 | xeet2 | junk-c: ahh, cool. but, how would I wind up using this information in an application? like, I want to say, if number 123 has 4 active calls, and a 5th comes in, only allow that call if 5 other channels are available, if not, issue a hangup or busy or whatever |
| 02:16.52 | JunK-C | I dont understand the question. |
| 02:16.52 | Dawson64 | /product_info.php?cPath=95_109&products_id=356 |
| 02:16.52 | Dawson64 | <shmaltz> $299 |
| 02:16.52 | Dawson64 | <xeet2> junk: is it an application or like a global variable? |
| 02:16.52 | Dawson64 | <JmanA9> happens with any fwd number |
| 02:16.52 | Dawson64 | <JunK-C> xeet2: http://bugs.digium.com |
| 02:17.06 | *** part/#asterisk Dawson64 (Dawson64@24.247.102.233.gha.mi.chartermi.net) |
| 02:17.09 | shmaltz | I know |
| 02:17.27 | FuriousGeorge | mgth: do all the ports on the tdm400p become wither fxo or fxs when using this module? or do multiple modules fit on one tdm400p |
| 02:18.08 | *** join/#asterisk cced (~wangxinta@222.33.36.198) |
| 02:18.08 | xeet2 | junk-c: we have a few pri's, we want to allow up to 4 channels for a given did number at any time, but only allow more than 4 channels for this did if there are more than 5 channels available on a given pri |
| 02:18.08 | mgth | a td400p has 4 ports on it....they do nothing until you put in a module |
| 02:18.17 | SPoon_TSX | Is PolyCOM have good sound quality? |
| 02:18.22 | JunK-C | if more then 4, refuse the call? |
| 02:18.41 | xeet2 | junk-c: if more than 4, an less than 5 open channels on the pri, refuse the call |
| 02:18.53 | FuriousGeorge | mgth: right, my question is do 4 modules fit on one tdm400p or does 1 module assign all the ports |
| 02:18.59 | xeet2 | but to make it simple yes, if more than 4, refuse the call |
| 02:19.14 | JunK-C | xeet2: use SetGroup/CheckGroup for that. |
| 02:19.29 | xeet2 | ahhh |
| 02:20.06 | JunK-C | my patch is just to know at a time t (present), how many calls are on a specific exten. |
| 02:23.14 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 02:23.27 | xeet2 | are there any restrictions on the "GROUP" value you set on setgroup? like, can I just do: 4445551212,1,SetGroup(4445551212) |
| 02:24.15 | SPoon_TSX | Hi Xeet2: I lookup the internet seems some people having some difficulties on updating the Phone ROM on Cisco 7960. Do you have such experience too? |
| 02:24.50 | *** join/#asterisk bonbon-home (~happy@81-86-0-190.dsl.pipex.com) |
| 02:25.03 | bonbon-home | guys, who can recommend me a decent rating engine? |
| 02:25.14 | xeet2 | spoon: no, everything worked fine, just have to follow the directions, have to upgrade the code in order |
| 02:25.24 | *** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com) |
| 02:25.38 | xeet2 | its not always just load new code, so its not something the average end user will be able to easily figure out |
| 02:26.03 | puppet | Anyone know where to get the cheapest possible analog<->digital card there is? modem is fine i just wanne lab with it |
| 02:26.05 | JunK-C | why not do 4445551212,1,SetGroup(${EXTEN}) then? |
| 02:26.39 | xeet2 | junk: that will work too, but I was wondering if there were limitations on the value of the group variable |
| 02:26.53 | SPoon_TSX | xeet2: What about the license, do I need license to use them on my Asterisk PBX network? |
| 02:27.25 | JunK-C | i dont think ca. |
| 02:27.31 | JunK-C | so |
| 02:27.36 | xeet2 | spoon: you will need to get the sip code, if it doesn't come with the phone you purchase. You can buy the code from cisco, or if you have a contract they'll usually give you access to get it yourself |
| 02:28.14 | hardwire | anybody have an older polycom videoconference phone? |
| 02:28.18 | xeet2 | sometimes you can get someone to send you the code, but thats not legal, so don't count on it |
| 02:28.22 | hardwire | I just wrote a joystick to telnet interface for it |
| 02:28.47 | xeet2 | puppet: what do you mean? just like a pots interface for a pc? |
| 02:28.59 | SPoon_TSX | oic. |
| 02:29.17 | SPoon_TSX | What about PolyCOM, do I need license on using their phone on my Asterisk Network? |
| 02:29.37 | *** join/#asterisk david (~dcoulson@muffin.davidcoulson.net) |
| 02:29.56 | *** part/#asterisk zignig (~simon@203.217.15.10) |
| 02:30.11 | xeet2 | you don't ever need a license to use any sip device with *, as long as the manufacturer doesn't put any restrictions on it, which usually they won't, since its an open standard |
| 02:30.48 | xeet2 | the only real license issues are getting the code for the device that can do sip |
| 02:30.55 | xeet2 | er, is |
| 02:30.56 | *** join/#asterisk Moc_ (~Moc@64.235.217.179) |
| 02:31.14 | xeet2 | junk: thanks for your help |
| 02:31.20 | JunK-C | xeet2: it works now? |
| 02:31.43 | xeet2 | junk-c: haven't tried it yet, but thats definitely the app I was looking for |
| 02:32.04 | JunK-C | i wonder if an app should be make for that |
| 02:32.14 | SPoon_TSX | oic. so is the Cisco phone I purchase off the voipsupply.com I can just plug into my Asterisk and will work without any additional purchase from Cisco? I am so dump on Licensing issue. |
| 02:32.21 | mgth | xeet2: if it is cisco you need a lic |
| 02:32.25 | xeet2 | junk-c: I'm sure some terminations provider that uses * has something written already =) |
| 02:32.49 | xeet2 | mgth: yes, like I said, you can buy it from cisco, and that if someone sends you the code its not legal |
| 02:33.24 | xeet2 | spoon: you need to find out what code the phone comes with. If its not sip, then you need to buy the sip code |
| 02:33.33 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 02:33.39 | xeet2 | or, use the code it comes with |
| 02:33.45 | xeet2 | that isn't sip |
| 02:33.45 | SPoon_TSX | xeet2: Do you know how much will it cost? |
| 02:34.13 | riksta | topping: sup |
| 02:34.14 | xeet2 | spoon: no idea, we bought a support contract that lets us get all the sip code we need, but that was about 1k |
| 02:34.32 | puppet | xeet2: yeah |
| 02:35.14 | xeet2 | puppet: you can get a rip off x100p on ebay, but the x100p from digium is designed for use with *, usually 100$ I believe |
| 02:35.31 | puppet | xeet2: ok |
| 02:35.38 | *** join/#asterisk BBRodriguez (~alex@pD9563F15.dip.t-dialin.net) |
| 02:35.44 | xeet2 | thats an fxs interface, you plug a pots line into it |
| 02:35.45 | puppet | omfg |
| 02:35.47 | BBRodriguez | Hi everyone |
| 02:35.47 | puppet | mt :) |
| 02:36.05 | BBRodriguez | Does anybody work with avop.net as a seller ? |
| 02:36.41 | SPoon_TSX | ok, thanks everyone. |
| 02:37.15 | xeet2 | is anyone here familiar with cisco 3600 hardware, specificly what cards are needed to connect to a pri and use sip? |
| 02:37.39 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 02:38.15 | FuriousGeorge | im having trouble understanding the logistics of this, if fxo/fxs modules can be daughterboards inside your computer, how do they allow anything to terminate into them? what kind of "port" is on the tdm400p? its not rj11, is it? |
| 02:39.04 | xeet2 | furious: its rj45, but rj11 fits into 45 |
| 02:39.11 | *** join/#asterisk ctooley (~ctooley@cpe-70-112-155-49.austin.res.rr.com) |
| 02:39.21 | FuriousGeorge | i see |
| 02:39.39 | xeet2 | furious: you use either fxo or fxs daughter boards on the card that connect directly to the 4 rj45/rj11 compatible ports on the back of the card |
| 02:39.51 | FuriousGeorge | so i put the module inside, and i connect either an rj11 or rj45 to the thing |
| 02:40.03 | BBRodriguez | Does anybody work with avop.net as a seller,using asterisk ? did anyone worked with voipescrow.net using asterisk as terminating GW ? |
| 02:40.29 | xeet2 | furious: yes |
| 02:40.51 | puppet | xeet2: http://www.digitnetworks.com/store/product_info.php?products_id=28&osCsid=537b1d19a45fbc28596abdbb444cf5b3 ;D |
| 02:41.04 | FuriousGeorge | xeet2: but the rj45 would be for an ip phone, not a computer, right? a softphone would be used over the regular lan infrastructure, no? |
| 02:41.09 | xeet2 | furious: only uses two pins so rj11 is sufficient. they just used rj45 so the plate would be the same for the 4 port t1/e1 cards |
| 02:41.18 | xeet2 | furious: no, the rj45 ports are analog |
| 02:41.47 | xeet2 | furious: you use your ethernet nic on your * box to connect to your lan that would have ip phones on it |
| 02:42.10 | FuriousGeorge | so what type of analog devices use an rj45 connection? |
| 02:42.37 | xeet2 | puppet: yeah, thats one of the clones. I personally would rather give digium the money for a real x100p |
| 02:42.48 | puppet | xeet2: whats the diff then? |
| 02:43.29 | xeet2 | furious: if you put an fxs module on the card for that port, then you connect a phone line from your telephone company to it. If you put an fxo module on the card for that port, then you can plug in any phone device, like a fax machine, regular phone, cordless phone, etc |
| 02:44.01 | puppet | xeet2: http://www.digium.com/index.php?menu=devkit-fxofxs |
| 02:44.02 | Luhiwu | xeet2, isn't it the other way? fxo to telco and fxs to personal devices? |
| 02:44.26 | puppet | So i cant connect a fax to my Cisco ata and route it threw * ? |
| 02:44.37 | puppet | fxo = pots |
| 02:44.41 | puppet | fxs = tele |
| 02:44.45 | xeet2 | I may have what digium calls their cards reversed. fxs signalled ports have to plug into a pots line, fxo signalled ports have to plug into a telephone device |
| 02:45.17 | Luhiwu | fxo port needs fxs signalling and fxs port needs fxo signalling |
| 02:45.20 | xeet2 | so fxs signalled ports is expecting to connect to another device that is fxo signalled, and vice versa |
| 02:45.20 | FuriousGeorge | i got that much. im new to telephony in general, but im gathering some phones use rj45, so im wondering what kind of phones they are, since business wiring is done w/ cat3 |
| 02:45.48 | xeet2 | furious: if you are using an ip phone, it will be rj45, standard ethernet |
| 02:46.00 | xeet2 | which you connect to your lan switch or whatever you're using, not directly to your * box |
| 02:46.02 | puppet | xeet2: Hmm but cant i use the clone to connect to a fax then? :o |
| 02:46.11 | *** join/#asterisk Moc_ (~Moc@64.235.217.179) |
| 02:46.52 | xeet2 | puppet: you can use an ata device for a fax machine, just use ulaw and put it on the local lan and you should be fine |
| 02:47.11 | FuriousGeorge | xeet2: we are having a communication break down and i'm sure its all my fault. i guess what im trying to ask is: why does the tdm400p have an rj45 connection? is that for external fxo fxs? |
| 02:47.20 | puppet | xeet2: that was my plan with the fax, my plan with the analog was simply to route the parents phone to min |
| 02:48.03 | xeet2 | furious: the tdm400p has rj45 because it is identical, faceplate wise, as the 4 port t1/e1 cards they make, and saves digium money. |
| 02:48.14 | FuriousGeorge | thanks ;) |
| 02:48.30 | xeet2 | the ports are rj45, but it is still an analog port, usually with rj-11 |
| 02:49.01 | puppet | xeet2: so if im just gone use the analog to route in out with voice a clonecard is ok? |
| 02:49.23 | xeet2 | puppet: you can do that, just keep in mind that the x100p is only an fxs signalled card, meaning it will not provide dialtone |
| 02:49.53 | puppet | xeet2: meaning it will just be incoming? |
| 02:50.10 | JunK-C | xeet2: the X100P is deprecated anyways. |
| 02:50.11 | puppet | or you mean the real dialtone when phone rings? |
| 02:50.23 | xeet2 | the clones will work if you're tight on money, but with all the contribution digium has made to voip I prefer to pay them the few extra $ =P |
| 02:50.40 | puppet | xeet2: bad thing i dont have the extra $ :/ |
| 02:50.42 | Katty | tata (= |
| 02:51.54 | JunK-C | someone have any great link about PDD? |
| 02:51.58 | xeet2 | puppet: understandable... fxs signalled meaning it expects a dialtone from the line, you can receive calls in and place calls out, but it won't provide signalling to connect a fax machine or regular phone to it. the two ports are pass through so you can put another device on it, but the dialtone will still only be passed from the telco |
| 02:52.44 | BBRodriguez | has anyone tried www.voipescrow2.net ? |
| 02:53.21 | puppet | xeet2: so I can just plug in my * to the analog net and get my * to work as a connected phone to the net? |
| 02:54.39 | xeet2 | if by analog net you mean the telco pstn, then yes |
| 02:54.50 | puppet | xeet2: yeah thats what i mean :) |
| 02:56.03 | xeet2 | * can pick up inbound calls and do whatever you want with them, like send it to a fax machine on your ata, to an ip phone, or whatever, and * can take calls received from an ip phone or your ata and dial them out via that x100p |
| 02:56.47 | puppet | xeet2: thats what i wanne do |
| 02:57.18 | puppet | xeet2: wanne do like if parents shout "u got phone!!!!!" ;p then i lift one of my ip phones and dial an extensions to * that connects it |
| 02:58.29 | xeet2 | you can do it that way, or you can have * automatically call your ip phone when a call comes in on the x100p, and ifyou answer the ip phone, it will automatically pick up and connect you to the pots line |
| 02:58.46 | xeet2 | just make sure you set immediate=no =) |
| 02:59.26 | SexyKen | Hey guys, I get this over and over again when I get calls into my IVR that are transferred to the queue and the call is tried to be transferred to agents who aren't answering: |
| 02:59.26 | SexyKen | 10pm] <|Vulture|> Ill have to try it tonight |
| 02:59.26 | SexyKen | [6:10pm] <|Vulture|> its a bit more complicated |
| 02:59.26 | SexyKen | [6:11pm] <SexyKen> So you think the logging looks okay? |
| 02:59.26 | SexyKen | [6:11pm] <|Vulture|> yup |
| 02:59.27 | SexyKen | [6:12pm] <SexyKen> Now when it says 798 calls is that 798 actual calls? |
| 02:59.29 | SexyKen | [6:12p |
| 02:59.31 | SexyKen | Fuck. |
| 02:59.33 | SexyKen | Never fucking mind. |
| 02:59.45 | SexyKen | 51. 2005-03-10 16:04:24 Local/3... 4176823322 4176823322 Dial SIP/1.303 303 NO ANSWER 00:16 |
| 02:59.45 | SexyKen | 52. 2005-03-10 16:04:05 Local/3... 8156936834 8156936834 Dial SIP/1.303 303 NO ANSWER 00:15 |
| 02:59.45 | SexyKen | 53. 2005-03-10 16:03:47 Local/3... 4176823322 4176823322 Dial SIP/1.303 303 NO ANSWER 00:15 |
| 03:00.04 | SexyKen | Is there anyway to link them together so we know that the call was from the same originating whatever you'd call it? |
| 03:00.08 | puppet | xeet2: :) |
| 03:00.11 | JunK-C | SexyKen: why are ya flooding? |
| 03:02.24 | puppet | sexyken: use pastebin.ca |
| 03:02.31 | puppet | sexyken: dont flood channel |
| 03:03.00 | *** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar) |
| 03:05.07 | SexyKen | Okay, then go here: http://pastebin.ca/7216 |
| 03:05.07 | JunK-C | ~seen mustdie |
| 03:05.13 | jbot | mustdie <~voip@ool-18b91fa7.dyn.optonline.net> was last seen on IRC in channel #asterisk, 114d 11h 3m 17s ago, saying: '~seen JerJer'. |
| 03:05.13 | *** join/#asterisk outsidefactor (~blah@203-173-32-225.dyn.iinet.net.au) |
| 03:05.13 | SexyKen | That's an example of my log. |
| 03:05.48 | JunK-C | what kind of log is that shit? |
| 03:05.49 | SexyKen | The user never hung up but it still inserts many records into the database, but it doesn't have any type of id that links the rows to the originating call. |
| 03:06.01 | SexyKen | That's mySQL buddy. |
| 03:06.04 | SexyKen | cdr_mysql |
| 03:06.11 | JunK-C | use .csv dude. |
| 03:07.17 | *** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.rr.com) |
| 03:09.24 | Nugget | mysql blows goats. |
| 03:09.29 | SexyKen | Yea -- and csv is better. |
| 03:10.23 | JunK-C | i love psql! |
| 03:10.29 | modulus_ | no you don't |
| 03:10.32 | modulus_ | you only think you do |
| 03:10.36 | *** join/#asterisk dontmsgme (~none@207.215.252.80) |
| 03:10.42 | dontmsgme | What is the syntax to register a DID from Nufone |
| 03:10.53 | modulus_ | register.did.nufone.now() |
| 03:10.57 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 03:11.04 | dontmsgme | isnt it Register => login:pw@switch-2.nufone.net ? |
| 03:11.08 | modulus_ | for all you OOP morons |
| 03:11.09 | JunK-C | modulus_: mouyhahaha |
| 03:11.15 | dontmsgme | Is that login/passowrd the same as the website login? |
| 03:11.33 | JunK-C | dontmsgme: go read on www.voip-info.org |
| 03:12.36 | dontmsgme | * Added junk-c!*@* to ignore list |
| 03:12.44 | puppet | lol @ dont |
| 03:12.59 | puppet | dontmsgme: go read instead of just asking,if you read you maybe learn something? :) |
| 03:13.04 | JunK-C | mouhahhaa |
| 03:13.34 | JunK-C | and he didnt say a simple hello before asking his question. :P |
| 03:13.39 | puppet | junk-c: no lol |
| 03:13.50 | puppet | junk-c: i bet he have removed ignore to now, just to see what u say ;p |
| 03:14.04 | JunK-C | hehe |
| 03:14.37 | puppet | junk-c: why use nufone btw? :) they cost cash |
| 03:14.57 | riksta | oh god it's dontmsgme |
| 03:14.58 | JunK-C | ive type nufone on voip-info |
| 03:15.03 | JunK-C | 2 clicks and i was on http://www.voip-info.org/wiki-Asterisk+settings+nufone |
| 03:15.13 | JunK-C | riksta: ? |
| 03:15.14 | puppet | junk-c: ipkall -> fwd |
| 03:15.19 | puppet | junk-c: works nice and its free ;p |
| 03:15.20 | dontmsgme | You'rwe right my question was too general but the simple answer of "google it" is stupid IMO |
| 03:15.30 | puppet | dontmsgme: NO Its nots |
| 03:15.31 | puppet | not* |
| 03:15.33 | riksta | that's because dontmsgme is an idiot (I am recalling from previous nights watching his babbel) |
| 03:15.39 | Eight | Does 1.0.6 resolve the issue fixed by the patch mentioned at the bottom of the Broadvoice-settings wiki page? http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice |
| 03:15.50 | dontmsgme | * Added riksta!*@* to ignore list |
| 03:15.57 | riksta | thank god for that |
| 03:16.10 | dontmsgme | Im like the Ghandi of IRC, silent protest |
| 03:16.17 | puppet | dontmsgme: http://images.google.se/images?q=google+bart&hl=sv&btnG=Google-s%C3%B6kning |
| 03:16.29 | puppet | riksta: yeah i did an last |
| 03:16.32 | puppet | riksta: was alot of junk |
| 03:16.34 | JunK-C | MOUHAHAHA |
| 03:16.43 | riksta | puppet: :) |
| 03:17.16 | puppet | (07:32:45) < JerJer[mobile]> dontmsgme: cuz you are smokin crack? |
| 03:17.21 | puppet | lol |
| 03:17.23 | puppet | hahhahahahaa |
| 03:17.26 | JunK-C | MOUHASHAHAHAHA |
| 03:17.29 | JunK-C | hehhee |
| 03:17.31 | JunK-C | i love that guy! |
| 03:17.33 | JunK-C | mouahha |
| 03:17.44 | puppet | what about this quote |
| 03:17.45 | puppet | (07:14:41) < dontmsgme> Does ssh need to be running to simply connect from windows with Xlite to the linux box? |
| 03:18.00 | riksta | i rest my case |
| 03:18.02 | JunK-C | is that a joke? |
| 03:18.08 | puppet | real quote |
| 03:18.20 | JunK-C | lol |
| 03:18.23 | puppet | (07:21:54) < dontmsgme> Is xlite made for FC2? |
| 03:18.33 | riksta | aww you embarassed him |
| 03:18.36 | puppet | hmm he left :( |
| 03:18.40 | puppet | no fun now :( |
| 03:18.41 | JunK-C | hehehe, who's that guy ? |
| 03:18.42 | riksta | SHAMED OUT! |
| 03:18.48 | riksta | he's a fuckin idiot |
| 03:18.55 | *** join/#asterisk cero64 (ruiner@fantab.ulo.us) |
| 03:18.58 | JunK-C | a fucking crack smoker :P |
| 03:19.01 | JunK-C | like jerjer said. |
| 03:19.14 | riksta | that's some strong crack |
| 03:19.15 | dan2 | JunK-C: just an idiot yea higher than bush |
| 03:19.31 | JunK-C | less dangerous. |
| 03:19.36 | JunK-C | until he find a job at BELL :P |
| 03:19.47 | puppet | http://pastebin.ca/7217 |
| 03:19.59 | riksta | he does the same thing as bush too, when someone opposes him, he just ignores em :) |
| 03:20.08 | ctooley | There needs to be a summary at the end of make that tells you all the decisions that were made and the options that were either followed or not followed. I just spent an hour trying to figure out what was wrong with my unixODBC configuration only to realize that unixODBC-devel wasn't installed so 'make' didn't build odbc support into Asterisk. |
| 03:20.23 | puppet | ctooley: sucks :/ |
| 03:20.31 | JunK-C | puppet: create him a wiki page for memory :P |
| 03:20.37 | puppet | ctooley: my meetme didnt compile cause i didnt have zap in ;P but i didnt know |
| 03:20.42 | puppet | junk-c: yeah took never expire to |
| 03:20.43 | puppet | junk-c: lol |
| 03:20.59 | riksta | * IMPORTANT: 36 config files in /etc need updating |
| 03:21.02 | riksta | god damn i hate gentoo |
| 03:21.11 | riksta | i'm too impatient to do repetitive tasks like that |
| 03:21.16 | Eight | Does 1.0.6 resolve the issue fixed by the patch mentioned at the bottom of the Broadvoice-settings wiki page? http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice |
| 03:21.16 | JunK-C | (06:13:59) < dontmsgme> How do you disable NAT |
| 03:21.18 | JunK-C | mouhahah |
| 03:21.32 | Eight | Or, alternatively, does it generate a 403 forbidden when i try to make an outgoing calling? |
| 03:21.40 | puppet | junk-c: ;p see fun? |
| 03:22.07 | riksta | 015 (07:11:51) < dontmsgme> Why can I ping my asterisk box from windows but I cant telnet in? |
| 03:22.14 | riksta | telnet!!?!?! |
| 03:22.37 | puppet | haha |
| 03:24.50 | riksta | ooh this new version of amarok is damn kool |
| 03:25.39 | JunK-C | http://stef.istop.com/ctu.wav |
| 03:27.27 | *** join/#asterisk SplasPood (jwb@paravolve.net) |
| 03:30.49 | JunK-C | [22:30:04] * ChanServ has quit IRC (Killed by alindeman ()) |
| 03:32.38 | Eight | Is this the error associated with not having the 'broadvoice patch' applied? chan_sip.c:6829 handle_response: Forbidden - wrong password on authentication for INVITE to '"Blake" <sip:##########@sip.broadvoice.com>;tag=as187d8755' |
| 03:34.09 | SexyKen | What is the disposittion field in asterisk for |
| 03:34.14 | SexyKen | In asterisk cdr |
| 03:36.21 | *** join/#asterisk beardedyak (~idhyc@0-2pool69-34.nas2.chattanooga2.tn.us.da.qwest.net) |
| 03:36.28 | JunK-C | SexyKen: go read about it on www.voip-info.org |
| 03:36.37 | SexyKen | I haven't seen anything about it. |
| 03:36.41 | SexyKen | Sweetie. |
| 03:36.41 | JunK-C | if its the call was ANSWERED,BUSY or NO ANSWER. |
| 03:36.53 | SexyKen | Is it actually accurate? |
| 03:37.03 | JunK-C | accurate means? |
| 03:37.15 | Eight | Alright. I'm thinking my BV account just isn't fully activated... if I call the number from a landline I get a "The party you tried is busy" recording... my account configs are setup for voicemail. |
| 03:37.47 | SexyKen | Wow, you dont know what accurate means? |
| 03:38.15 | JunK-C | SexyKen: nope. |
| 03:38.31 | riksta | SexyKen: accurate doesn't make sense in that context |
| 03:38.31 | JunK-C | but i know whats the disposition field :) |
| 03:38.38 | riksta | hahahah JunK-C :) |
| 03:38.42 | JunK-C | hehhee |
| 03:38.48 | riksta | you are on form tonight |
| 03:38.59 | Eight | oh, whoa... my bad. I just didn't listen to the whole message =p |
| 03:39.06 | JunK-C | so tell me what accurates means i'll tell ya what's the disposition :) |
| 03:39.14 | JunK-C | riksta: ya, im taking it easy 2nite. |
| 03:39.25 | riksta | :) |
| 03:39.41 | SexyKen | Yes it does, riksta. I'm asking if the data in the field is accurate. |
| 03:39.44 | JunK-C | and we started a lot of agis at the same time, no miss call, so im a bit happy. |
| 03:39.52 | riksta | any of you bitches wanna help on my ADM project, for asterisk |
| 03:39.52 | JunK-C | if its important? |
| 03:39.54 | *** join/#asterisk wasim (~wasim@203.81.217.160) |
| 03:40.06 | riksta | SexyKen: that STILL doesn't make sense |
| 03:40.22 | SexyKen | Then you're stupid. |
| 03:40.27 | riksta | no son, you are |
| 03:40.35 | SexyKen | Not at all. |
| 03:40.38 | puppet | is it good/bad to have a macro call a macro? |
| 03:40.43 | riksta | structure your question correctly |
| 03:40.47 | SexyKen | Anyone who understands english would comprehend the question. |
| 03:41.02 | riksta | would that be obfuscated english? |
| 03:41.31 | SexyKen | Do you know what accurate means? |
| 03:41.41 | Nugget | many people do not speak english as their first language, Ken. |
| 03:41.47 | Nugget | how's your swahili? |
| 03:42.07 | SexyKen | Nugget -> The point is I dont correct someone when they speak swahili to me. I dont tell them their swahili is wrong. |
| 03:42.15 | puppet | accurate ~exact |
| 03:42.15 | riksta | SexyKen: yes. |
| 03:42.31 | puppet | =~ exact * |
| 03:42.39 | JunK-C | swahili is a language? i though it was an animal :P |
| 03:42.42 | JunK-C | kmouhahaha |
| 03:42.46 | Nugget | heh |
| 03:42.58 | SexyKen | without error, precise, exactness to a specific degree or size, true to given dimensions. |
| 03:43.12 | riksta | no SHIT |
| 03:43.34 | SexyKen | And you're gay ass argument is that the question isn't phrased proprely? |
| 03:43.43 | SexyKen | That it doesn't make sense? |
| 03:44.02 | Eight | Guys... could you just drop this? |
| 03:44.05 | Eight | It's obviously not productive. |
| 03:44.17 | Eight | If you want to keep bitching at eachother, how about using another channel for it? |
| 03:44.24 | SexyKen | CDR is FAR from perfect. I have a doubt in my mind that it may report invalid information to the DB, especially considering past results. |
| 03:44.25 | moonwick | "you're gay ass argument" just beats me over the head with its eloquence. |
| 03:44.36 | SexyKen | *your |
| 03:46.08 | SexyKen | So I suppose if I was stupid enough to think that Asterisk, the open source PBX, was a perfect system with no problems at all, the question wouldn't make sense. |
| 03:46.35 | SexyKen | Anyway...off to figuring out why CDR is reporting wrong data. |
| 03:47.54 | beardedyak | Newb here... anyone open to providing a hardware recommendation for a particular try-out situation, or am I too far off-topic? |
| 03:48.02 | JunK-C | SexyKen: whats wrong? |
| 03:48.40 | SexyKen | I suppose that the data it's reporting isn't exactly wrong, just not organized in a very useful way. |
| 03:48.42 | Eight | beardedyak: You want to try out Asterisk? |
| 03:48.54 | beardedyak | I'm mainly interested in trying asterisk for the first time and demonstrating the basic functionality to my boss |
| 03:49.00 | Eight | beardedyak: Are you talking about the host machine, or hardware to interface with the PSTN? |
| 03:49.02 | beardedyak | I am the LAN/WAN admin |
| 03:49.04 | JunK-C | SexyKen: which data exactly? any specific field? |
| 03:49.34 | Eight | beardedyak: For a simple demo just about any hardware you can install a modern distro on will be plenty. |
| 03:49.45 | SexyKen | Assume someone calls an 800# and enters a context with my IVR in it. They enter 1 to join the 'Sales Queue'. Once they're in the queue Asterisk sees an 'available' agent, so it tries to connect the call. The agent doesn't answer. This happens 15 times. |
| 03:49.56 | Eight | beardedyak: And if you like you can just use soft phones for the demo (Xten's X-Lite, for example). |
| 03:49.59 | SexyKen | Therefore, Asterisk logs 15 separate calls. I need a way to link them together. |
| 03:50.03 | Eight | beardedyak: which means you won't need to buy any hardware. |
| 03:50.24 | Eight | beardedyak: otherwise Digiums TDM400P card with a couple FXS and FXO ports are what you want to talk to normal analog phones and lines. |
| 03:50.28 | beardedyak | Eight: I was thinking if I could maybe hook up a single POTS line to an asterisk system and maybe use a WiFi phone with it to start with, but soft phones sound like a better beginning |
| 03:50.38 | SexyKen | I want to be able to track the call from the second it calls the 800# to the second the call was answered and hung up by an actual agent or whatever the last even was. |
| 03:50.44 | SexyKen | *event |
| 03:51.12 | *** join/#asterisk mrproper_ (~psynode@CPE-203-45-98-127.nsw.bigpond.net.au) |
| 03:51.28 | Eight | beardedyak: You can atleast try it out with only SIP, then add the TDM400P card when you want to explore that. |
| 03:51.34 | mrproper_ | anyone here heard of getting microsoft live communications server working with asterisk? |
| 03:51.43 | Eight | Busy?! I'm not BUSY! |
| 03:52.00 | beardedyak | Eight: thanx for the suggestions. Is it pretty easy to set this all up, or will I be hanging out in IRC a lot during this? |
| 03:52.16 | *** join/#asterisk jeffik (~jeffik@user-v7kajsk.dialup.mindspring.com) |
| 03:52.24 | Eight | beardedyak: voip-info.org is your friend |
| 03:52.27 | Eight | jbot wiki? |
| 03:52.28 | jbot | i heard wiki is http://www.voip-info.org |
| 03:52.51 | Eight | Hey, got it on the first try this time. |
| 03:53.01 | SexyKen | Junk-C, does this make any sense to you? |
| 03:53.26 | Eight | beardedyak: Familiar with Sendmail? |
| 03:53.56 | beardedyak | Eight: yes to sendmail. little rusty, tho |
| 03:54.27 | Eight | beardedyak: I figure Asterisk is about equivalent as far as learning curve. |
| 03:54.57 | Eight | beardedyak: not that i"m particularly far along that curve myself. |
| 03:55.00 | JunK-C | SexyKen: i cant say, ive not touched agents and queues atm |
| 03:56.06 | SexyKen | I'm thinking I may be able to use SetGlobalVar() and SetUserField() to store a unique 'call-tracking' id...but I'm just not sure how I'd do it |
| 03:56.41 | Eight | Re: Broadvoice and account setup: Is it normal to have both incoming and outgoing callings with Broadvoice hit a 'busy' message before the account is fully setup? |
| 03:57.04 | Eight | Or once I get that far, is it active and the fact that it doesn't work is my fault? =/ |
| 03:57.19 | beardedyak | Eight: I think I'll be okay, then. Sendmail is a bit of a labor of love, but has a certain enjoyable logic. |
| 03:57.22 | xeet2 | eight: that usually means something isn't configured correctly |
| 03:58.00 | xeet2 | sexyken: yeah you should be able to do that, you can set a user field to be *'s unique call id, I forget what the variable name is though |
| 03:58.12 | Eight | xeet2: Ya, it's throwing a 'wrong pass' error when I try to make outgoing... |
| 03:58.33 | xeet2 | eight: you're using the other password they sent you, right? not the one you log in to the site with |
| 03:58.40 | Eight | xeet2: yup. |
| 03:58.46 | SexyKen | xeet, Are you speaking of: ${UNIQUEID}: Current call unique identifier |
| 03:59.04 | xeet2 | yes |
| 03:59.08 | SexyKen | k. |
| 03:59.10 | SexyKen | Thanks :-) |
| 03:59.23 | xeet2 | make sure you grab that before it goes to a call queue though |
| 03:59.47 | SexyKen | Oh you mean it'll change? |
| 03:59.56 | *** part/#asterisk ctooley (~ctooley@cpe-70-112-155-49.austin.res.rr.com) |
| 03:59.59 | beardedyak | eight: thanks again for helping me get started with your suggestions! bye! |
| 04:00.15 | Mneumonic | ManxPower - u there? |
| 04:00.26 | SexyKen | xeet, The Unique ID will change when it goes to the call queue? |
| 04:00.35 | xeet2 | I believe the value of that variable will change to be the call id of the call created by the queue, ie one of your 15 unique ones |
| 04:01.07 | xeet2 | so you should set a userfield to that variable when the call comes in, then it should pass it to all 15 of the other calls |
| 04:01.15 | Mneumonic | anyone know why i dont have chan_alsa.so in my modules dir? or how to get it? |
| 04:01.17 | xeet2 | as the userfield, not the unique id |
| 04:01.49 | xeet2 | don't quote me on that though, I only mess with call queues when I have to |
| 04:01.52 | puppet | anyone that have ast-real-time.rar? |
| 04:01.53 | SexyKen | xeet2, I understand. So basically, I want to get the unqiue call id the second the call gets to asterisk, then set a global variable with the unique id....and store it each time the call is xfer'd? |
| 04:02.15 | xeet2 | I wouldn't use a global variable, use the userfield |
| 04:02.30 | xeet2 | but essentially |
| 04:02.56 | xeet2 | I'm sure a global variable would work too, just have to make sure the variable name would be unique =) |
| 04:03.15 | xeet2 | and make sure it gets into the cdr record |
| 04:03.17 | SexyKen | xeet, Wont I still have to use a global variable in asterisk to store the id as it gets passed around different agents/attempts to connect to agents? |
| 04:03.55 | xeet2 | I thought userfields carried over to queues |
| 04:04.27 | xeet2 | the unique id has to change however, because * has another unique call to deal with |
| 04:05.28 | SexyKen | Ah, okay I didn't know userfields were carried over. |
| 04:06.07 | xeet2 | again, don't quote me on that, I thought they did |
| 04:06.17 | xeet2 | if they don't, then yes a global variable would be a way to do it |
| 04:07.03 | xeet2 | they get carried over for calls that come in and then dial out elsewhere, to the outbound channel, so its only natural |
| 04:10.37 | SexyKen | xeet2, It's odd right now....some calls will get a uniqueid and some wont. |
| 04:10.58 | xeet2 | do you see any pattern to the ones that do and the ones that don't? |
| 04:11.27 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 04:13.43 | Mneumonic | anyone know how to make it so that when the auto attendant picks up and the calling party doesnt hit any buttons for say 8 secons to ring a operators extension? |
| 04:14.19 | shido6 | yesh |
| 04:14.35 | Mneumonic | hey shido |
| 04:14.37 | Nugget | hang it off the "t" extension. |
| 04:14.39 | shido6 | set a digitimeout and responsetimeout and set a exten => t,1,Dial bleh |
| 04:14.43 | shido6 | so when it reached 8 |
| 04:14.44 | SexyKen | xeet, Actually I'm wrong. They all have a unique id except for the first 10...which was before I recompiled the module. |
| 04:14.44 | shido6 | seconds |
| 04:14.46 | shido6 | BEWM |
| 04:14.50 | shido6 | t gets exec'td |
| 04:14.52 | shido6 | -t |
| 04:15.37 | shido6 | [CC] |
| 04:15.37 | shido6 | exten => s,1,Answer |
| 04:15.37 | shido6 | exten => s,2,DigitTimeout,8 |
| 04:15.37 | shido6 | exten => s,3,ResponseTimeout,8 |
| 04:15.37 | shido6 | exten => s,4,BackGround( |
| 04:15.40 | shido6 | bleh) |
| 04:15.46 | SexyKen | xeet, So what would the process be to setting the userfield to an identifier and carrying it over to the queues and cal transfers? |
| 04:16.02 | shido6 | exten => t,1,Dial(SIP/STUPIDOPERATORMAN||r) |
| 04:16.21 | shido6 | or something like exten => t,1,Playback,stdbywhileitransferyourasstoanoperator |
| 04:16.31 | shido6 | exten => t,2,Dial(SIP/stupidoperatordood||r) |
| 04:17.04 | SexyKen | shido what is that for? |
| 04:17.07 | Mneumonic | i can use a Goto(100,1) on that too, right? |
| 04:17.27 | Nugget | of course. |
| 04:17.31 | Nugget | it's just like any other extension |
| 04:17.55 | eipi | why i get Unable to create channel of type 'IAX2' on iaxy disconnected phone? how * can detect that iaxy is disconnected and return NO-ANSWER? |
| 04:18.40 | Mneumonic | thanx guys |
| 04:18.54 | shido6 | dagnabbit |
| 04:19.02 | shido6 | are you using friend for the iaxy or a user and a peer in iax.conf |
| 04:19.02 | shido6 | ? |
| 04:19.11 | eipi | firend |
| 04:19.14 | shido6 | why? |
| 04:19.16 | shido6 | dont use friends |
| 04:19.20 | shido6 | just dont do it :) |
| 04:19.24 | eipi | :D |
| 04:19.24 | shido6 | break the friend |
| 04:19.26 | shido6 | out to a user |
| 04:19.27 | shido6 | and peer |
| 04:19.31 | shido6 | and u wont have that problem |
| 04:19.45 | eipi | ok, and how i can unregister that client? |
| 04:19.54 | shido6 | unplug it |
| 04:20.11 | shido6 | IAXys dont have on and off buttons |
| 04:20.17 | eipi | shido6 the problem started when i unplugged |
| 04:21.08 | *** join/#asterisk orrious (~paul@pcp01944638pcs.canton01.mi.comcast.net) |
| 04:21.22 | shido6 | err k |
| 04:21.23 | shido6 | well |
| 04:21.27 | shido6 | make the user and peer foist |
| 04:21.30 | shido6 | and then work from there |
| 04:21.50 | *** join/#asterisk sariabod (~sariabod@c-24-126-21-237.we.client2.attbi.com) |
| 04:22.05 | orrious | hello all.. newbee to asterisk here.. have a couple questions if you guys don't mind answering |
| 04:22.22 | shido6 | shoot |
| 04:22.36 | orrious | you guys played with asterisk@home? |
| 04:22.54 | orrious | was looking to use it because from what i've read it has a web interface |
| 04:23.18 | orrious | does the normal distro of asterisk have a web interface or only asterisk@home? |
| 04:23.20 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 04:23.38 | shido6 | eerrrr |
| 04:23.39 | shido6 | dood |
| 04:23.40 | sariabod | can the password for bugs.digium.com be reset? |
| 04:23.48 | shido6 | u can add an interface to your own build of asterisk |
| 04:23.58 | shido6 | asterisk@home TRIES to stick everything on a live or bootable cd |
| 04:24.08 | orrious | yes, but why re-invent the wheel? |
| 04:24.12 | shido6 | u dont need to use that.. make your own install |
| 04:24.18 | shido6 | and configure it to do whatever the hell you want |
| 04:24.24 | orrious | i know, but that doesn't work worth a sh!t |
| 04:24.29 | shido6 | because the wheel you are using is square |
| 04:24.33 | orrious | lol |
| 04:25.38 | orrious | don't get me wrong.. I'm a Cisco CallManager guy by trade.. one of my buddies asked me to set this up for him.. and it looks pretty interesting.. I just don't feel like writing a bunch of code if there is already something decent out there |
| 04:26.15 | orrious | so keeping that in mind.. is there anything that someone can use that has little to no knowledge of linux.. |
| 04:27.01 | shido6 | yeah |
| 04:27.31 | adeilt | why does asterisk grab my audio hardware? is it required? can this behavoir be disabled? i'd like to use my soundcard for other things. |
| 04:27.46 | orrious | i would like to use RHE with a clean install of asterisk. and a nice interface |
| 04:28.03 | orrious | pref web so i don't have to kill this box running X |
| 04:30.08 | Nugget | asterisk has a nice interface. it has a console. |
| 04:30.21 | Nugget | if you want some gui thing you are going to be disappointed. |
| 04:30.31 | shido6 | err |
| 04:30.31 | shido6 | oss |
| 04:30.34 | shido6 | u can turn that off |
| 04:31.04 | orrious | i have yet to install it.. wanted to do a little research before jumping in. what is the standard interface like? some CLI or menu driven? |
| 04:31.08 | adeilt | oss... thanks ... i'll grep it out and kill it |
| 04:31.16 | Nugget | it's just a text console. |
| 04:31.29 | orrious | what packages are required for the install? |
| 04:31.30 | Nugget | all the heavy lifting is done by editing config files |
| 04:31.36 | orrious | just the min, |
| 04:31.45 | Nugget | it has minimal dependancies. |
| 04:31.58 | orrious | know where there is a list? |
| 04:32.03 | Nugget | nope |
| 04:32.35 | orrious | figures. when i install the asterisk package, will it complain if a package isn't installed? |
| 04:32.49 | Nugget | I've never tried to install a package, I wouldn't know. |
| 04:33.08 | *** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl) |
| 04:35.28 | shido6 | err |
| 04:35.36 | shido6 | install the asterisk package, eh? |
| 04:35.43 | opus___ | does anyone here use AMP? |
| 04:35.46 | shido6 | cvs head all the way man |
| 04:36.03 | shido6 | amp needs work but does work for small installs...sorta |
| 04:36.07 | orrious | take it there is only source |
| 04:36.10 | shido6 | stupid friends |
| 04:36.24 | opus___ | shido6 - on the default install is /panel suppose to be blank? |
| 04:36.50 | shido6 | I dont know what /panel is |
| 04:36.58 | opus___ | Flash panel <directory> |
| 04:37.29 | shido6 | amp has flash components now that gastman sorta interface |
| 04:37.34 | Nugget | no, it should load the flash thingey. |
| 04:37.36 | shido6 | its like a sub xmas tree |
| 04:38.03 | Nugget | http://suburbia.slacker.com/asterisk/ <-- a working flash panel install |
| 04:38.31 | opus___ | hmm. thanks |
| 04:38.58 | opus___ | I guess I need a [sip] entry, thanks |
| 04:39.34 | shmaltz | Nugget, why is ivo (203) greyed out? |
| 04:39.39 | Nugget | he's offline |
| 04:39.59 | opus___ | nugget - can you show me a few lines from op_buttons.cfg? |
| 04:40.25 | shmaltz | I never new it will do that. is this by default? |
| 04:40.35 | Nugget | yes |
| 04:40.37 | shmaltz | what version ? .20? |
| 04:40.51 | Nugget | http://pastebin.ca/7219 |
| 04:41.18 | Nugget | .19 still. |
| 04:41.21 | Nugget | I'm a slacker :) |
| 04:41.58 | shmaltz | I have version .20, but I never noticed the offline status. I guess b/c I never had a phone offline |
| 04:42.14 | shmaltz | I modified it to fit 66 buttons on a page |
| 04:42.15 | xeet2 | is 204 moose by chance someone with the last name shabat? |
| 04:42.20 | Nugget | no |
| 04:43.01 | *** join/#asterisk alexns (~alex@acs-24-154-114-15.zoominternet.net) |
| 04:43.28 | shmaltz | Nugget, whats the password? I want to ring some phones :) |
| 04:43.30 | shmaltz | No I dont expect an answer |
| 04:44.10 | Nugget | heh |
| 04:46.12 | puppet | nugget: can you show the in/out lines some off them? |
| 04:47.10 | Nugget | they're pretty much straight from the default config. |
| 04:47.36 | puppet | Hmm ok gone take a glance over default config again then Kinda irritating to just have sip/* ;p |
| 04:51.24 | eipi | why i get Unable to create channel of type 'IAX2' on iaxy disconnected phone? how * can detect that iaxy is disconnected and return NO-ANSWER or BUSY or another status but no error? |
| 04:51.53 | eipi | the phone is not disconnected sorry, the iaxy is unplugged |
| 04:52.29 | puppet | Hmm |
| 04:53.50 | shido6 | eipi |
| 04:53.57 | shido6 | what do u have in your iax.conf |
| 04:53.59 | shido6 | pastebin.ca |
| 04:54.16 | shido6 | did u do the peer and user |
| 04:54.19 | shido6 | as I suggested? |
| 04:54.50 | `Sauron | Nugglet |
| 04:55.26 | Eight | Work dammit. |
| 04:55.36 | `Sauron | Eight: still fighting BV? :) |
| 04:57.23 | Eight | `Sauron: ya. |
| 04:57.33 | Eight | I applied the patch from the wiki... still 403 on dial out. |
| 04:57.35 | `Sauron | man |
| 04:57.42 | `Sauron | still the invite auth problem? |
| 04:57.46 | Eight | yup |
| 04:57.53 | Eight | register seems to go fine. |
| 04:57.57 | `Sauron | on the voip-info wiki page, there's a link to a wiki on robotic.net |
| 04:58.00 | `Sauron | or whatever the site's called |
| 04:58.21 | `Sauron | cut+paste that sip.conf entry into your sip.conf, replace the appropriate stuff, and it should work |
| 04:58.28 | `Sauron | in latest -head it will, anyway |
| 04:58.44 | `Sauron | I did that last night |
| 04:58.44 | Eight | I'm using 1.0.6 + Wiki patch for 1.0.5... |
| 04:58.56 | `Sauron | hum, try what I said anyway :) |
| 04:59.15 | Eight | What site it is it on? I've tried a variety of configs. |
| 04:59.24 | Eight | They're all pretty much the same. |
| 04:59.26 | `Sauron | dunno |
| 04:59.33 | `Sauron | go to bv settings on voip-info |
| 04:59.45 | `Sauron | at the top, there's an URL to a diff. site |
| 04:59.48 | `Sauron | that's also a wiki |
| 04:59.54 | `Sauron | with a page called Broadvoice_Settings |
| 05:00.05 | `Sauron | http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup |
| 05:00.06 | `Sauron | err |
| 05:00.08 | Eight | got it. |
| 05:00.09 | `Sauron | that link right there :) |
| 05:00.23 | `Sauron | cut+paste the sip.bv.com entry |
| 05:00.31 | `Sauron | and modify all the places it says phoneno and passwd |
| 05:00.34 | `Sauron | reload |
| 05:00.39 | `Sauron | and see if it works |
| 05:00.45 | Eight | user=phone? |
| 05:00.57 | `Sauron | shrug, it works |
| 05:01.10 | opus___ | hmmm |
| 05:01.27 | `Sauron | that's the official unofficial way |
| 05:02.46 | Eight | `Sauron: no joy =/ |
| 05:02.52 | Eight | 403 |
| 05:02.52 | `Sauron | hum, bummer |
| 05:03.03 | `Sauron | did you get the email with your account info? |
| 05:03.05 | puppet | Nerf Voicemail :( You cant really use it since it mails WAV and not mp3 :( |
| 05:03.07 | opus___ | Hmmm still blank operator panel :( |
| 05:03.39 | Eight | `Sauron: yup. |
| 05:03.41 | alexns | eight, i have bv & i found out the order of stuff in sip config makes a difference |
| 05:03.48 | Eight | And I can login on their website. |
| 05:04.13 | Eight | alexns: funny you should mention that... I just (for no good reason, I thought) matched the order used on the link `Sauron sent me. |
| 05:04.43 | alexns | i went crazy with it, then i changed order & it worked |
| 05:05.06 | alexns | i just use head version no patch |
| 05:05.14 | `Sauron | yeah, it's in head |
| 05:05.30 | Eight | argh. |
| 05:05.38 | Eight | proxy.chi.broadvoice.com doesn't work. |
| 05:05.58 | Eight | sip.broadvoice.com does |
| 05:06.19 | Eight | I suspect they just haven't gotten the remote machine updated with my info at this point. |
| 05:07.10 | alexns | i couldn't get the proxys to work either |
| 05:07.34 | alexns | i read before that you have to proxy server in hosts file |
| 05:07.48 | `Sauron | sip.bv.com points at proxy.dch.bv.com |
| 05:07.56 | `Sauron | I think chi is permanently down |
| 05:08.03 | alexns | hehe |
| 05:08.04 | `Sauron | can't ping it, anyway |
| 05:08.18 | alexns | i use broadvoice cause you basically get unlimited channels |
| 05:08.20 | Eight | really? I ping it in 20ms |
| 05:08.42 | Eight | I'm 45ms from sip.broadvoice.com |
| 05:08.50 | loud | alexns, they say you only have ONE clear channel. |
| 05:08.58 | alexns | yeah yeah |
| 05:09.03 | moonwick | ewww, clearchannel. |
| 05:09.03 | loud | hah |
| 05:09.04 | *** join/#asterisk Mikrozaur (~mikrozaur@213.206.33.132) |
| 05:09.20 | Mikrozaur | Hi |
| 05:09.28 | AgiNamu | Hey, WTF, someone in -dev said it took them 3 weeks to implement CC charging their first tiem around.... |
| 05:09.34 | AgiNamu | What were they writing it in ? x86? |
| 05:09.38 | alexns | i've used 6 @ time on cable modem |
| 05:09.42 | alexns | its good |
| 05:09.47 | loud | cool |
| 05:10.02 | alexns | i'm sure i could do way more with 729 |
| 05:10.02 | loud | show me your bill next month :) |
| 05:10.13 | AgiNamu | I just wrote a full Asterisk CDR import, rating, billing, cc processing, account setup (create peers, ext. etc.), customer interface, management interface, in 1 month |
| 05:10.22 | shmaltz | is a 300+ ms time for ping good for VOIP? |
| 05:10.26 | `Sauron | no |
| 05:10.28 | AgiNamu | shamtlz, no |
| 05:10.29 | alexns | i was already billed, was not any different. |
| 05:10.31 | AgiNamu | it'll WORK |
| 05:10.35 | AgiNamu | but it'll have delay |
| 05:10.39 | AgiNamu | satellite? |
| 05:10.43 | shmaltz | What is considered good? |
| 05:11.13 | AgiNamu | under 200 or so |
| 05:11.14 | alexns | anyway how would you limit channel usage in asterisk ? |
| 05:11.25 | Essobi | Mmm. |
| 05:11.37 | shmaltz | I am using simpletelecom.com they have 310 - 330 ms on pings and it works very well, actualy clearer than my digium card |
| 05:11.51 | Essobi | AgiNamu CGI development isn't hard.. MySQL backend? |
| 05:12.10 | *** join/#asterisk buddah (~buddah@66-214-239-6.lb-cres.charterpipeline.net) |
| 05:12.10 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
| 05:12.18 | Eight | I did the hosts trick, still bails. |
| 05:12.31 | AgiNamu | Essobi - MySQL LOL yea right |
| 05:12.33 | puppet | eight: problem with provider? :/ |
| 05:12.34 | AgiNamu | I'm using a REAL DB |
| 05:12.39 | AgiNamu | Everything is C#, + ASP.NET |
| 05:12.41 | opus___ | alexns - whats 729 |
| 05:12.44 | AgiNamu | even the parts that run on asterisk |
| 05:12.54 | AgiNamu | That's why it only took a month. VS2005 lets me fly |
| 05:12.58 | alexns | other codec in asterisk |
| 05:13.06 | Eight | puppet: I can get it to work fine with the main sip.broadvoice.com server, but the chicago proxy (which is half the 'distance' from me) just throw 403. |
| 05:13.06 | alexns | you gotta buy, 10$ per channel |
| 05:13.11 | AgiNamu | I dont try mysql for transactions correctly |
| 05:13.16 | alexns | hmm |
| 05:13.16 | puppet | eight: :( |
| 05:13.23 | AgiNamu | and on top of that, some of the billing stored procedures are complex |
| 05:13.26 | Eight | puppet: but it works, at all, so I'm happy. |
| 05:13.37 | AgiNamu | That said, when we commercialize it, I might write other backends |
| 05:13.38 | puppet | eight: glad I have my nice provider that dont limit number off in/out channels ;p |
| 05:13.39 | Essobi | AgiNamu You poor poor bastard.. ASP and C sharp. |
| 05:13.41 | AgiNamu | but I doubt it |
| 05:13.44 | shepherd | agi: stored procedures are fluff :) |
| 05:13.48 | AgiNamu | SQL 2005 Express is free |
| 05:13.51 | Mikrozaur | Can i do use soft sip dialer (x-lite) for use confirence on asterisk? I'm not have any special hardware. |
| 05:13.57 | Eight | puppet: I don't see how that applies... exactly. |
| 05:14.00 | AgiNamu | oh, i can do it without a sproc |
| 05:14.02 | Essobi | AgiNamu I could out code you in PHP and MySQL. :) |
| 05:14.06 | puppet | aginamu: the sql express that had a big sploit? ;P |
| 05:14.14 | AgiNamu | puppet ? |
| 05:14.23 | opus___ | MDAC |
| 05:14.27 | AgiNamu | But bringing 1000+ resultsets down to the app, processing, then sending them back, inside a repeatable read transaction |
| 05:14.32 | puppet | aginamu: wasnt there one sql lite version that had a huge sploit some months ago? |
| 05:14.33 | AgiNamu | doesn't sound like a good idea to me. |
| 05:14.36 | AgiNamu | SqlLite? |
| 05:14.42 | AgiNamu | I dunno. This is Microsoft SQL Server Express |
| 05:14.50 | AgiNamu | 2GB RAM, 2CPU, 4GB DB, $0 |
| 05:14.51 | Essobi | Eww. |
| 05:14.55 | AgiNamu | 2005 |
| 05:15.05 | Nugget | nothing wrong with mssql. it's a perfectly solid database. |
| 05:15.14 | Essobi | I've got CDR tables with over 28 million records in it. |
| 05:15.15 | puppet | nugget: there is better tho :) |
| 05:15.16 | Nugget | it's a fuckload better than mysql, that's for sure. |
| 05:15.17 | Essobi | :) |
| 05:15.20 | Essobi | Maha. |
| 05:15.23 | AgiNamu | Oracle, MS SQL... yea |
| 05:15.26 | AgiNamu | maybe DB2 |
| 05:15.27 | puppet | nugget: f.ex oracle :) |
| 05:15.31 | Eight | Hmm, anyone know a good way to fake a dial tone? I can do Playtones(dial) but it doesn't stop when the user starts to dial an ext'. |
| 05:15.35 | Essobi | My.. opinionated ehh? |
| 05:15.37 | Nugget | oracle and mssql aren't really competitors. |
| 05:15.40 | puppet | oracle > mysql5 > mssql > mysql |
| 05:15.41 | puppet | ;D |
| 05:15.55 | AgiNamu | although, my friend who works at IBM can't stop bitching about IBM's shitty products. ASked to work for free on some VS.NEt projects for me :P |
| 05:15.56 | Nugget | I wouldn't suggest oracle anyplace where mssql makes sense, and vice versa |
| 05:16.00 | opus___ | where does op_panel's flash applet look for its configuration file? |
| 05:16.05 | shepherd | oracle is a real db |
| 05:16.08 | Essobi | select count(*) where user > MySQL. |
| 05:16.09 | shepherd | mssql is just crap |
| 05:16.10 | Nugget | opus___: it talks to the op_server daemon |
| 05:16.14 | Essobi | mssql is shite. |
| 05:16.15 | AgiNamu | Nugget, well, having them compete, and making sense are different things. |
| 05:16.16 | Nugget | mssql is not crap. |
| 05:16.17 | puppet | opus___: same dir as u run the server |
| 05:16.33 | Nugget | it's not oracle, but it's not crap either |
| 05:16.34 | opus___ | oh its a deamon |
| 05:16.35 | shepherd | nugget: after you use oracle.. it is |
| 05:16.42 | Nugget | oracle does different things. |
| 05:16.45 | Essobi | unless it's on 64bit, and even then.. it's still not as fast as a linux/Bsd box running mysql in raw record throughput. |
| 05:16.46 | Nugget | I use both. |
| 05:16.47 | puppet | opus___: the op_server.pl is |
| 05:16.48 | shepherd | oracle does everything :) |
| 05:16.49 | AgiNamu | I went to one company, gave them a quote for an intranet (that had to integrate to Office). They said no. Why? It had to run on their $,$$$,$$$ Solaris SuperDuperFlareSparc system |
| 05:16.52 | SexyKen | Someone wanna help me set something in the Userfield? I'm having a problem. I try set it on a incoming did. |
| 05:17.00 | AgiNamu | because they overpurchases, and were only using 3% of the system |
| 05:17.04 | AgiNamu | even with a 500GB database |
| 05:17.14 | AgiNamu | Had I showed up with a super expensive Oracle app, they'd have bought it |
| 05:17.21 | Nugget | sometimes doing everything is a downside. :) |
| 05:17.37 | Essobi | Shit happens.. you take care of the customer, or else someone else will. |
| 05:17.40 | AgiNamu | All because the boss wanted to see the usage % up |
| 05:17.45 | *** join/#asterisk cero64 (ruiner@fantab.ulo.us) |
| 05:17.49 | SexyKen | exten => 8666666666,1,SetCDRUserField(${UNIQUEID}) |
| 05:17.49 | SexyKen | exten => 8666666666,2,Goto(1.incoming-ivr,s,1) |
| 05:17.50 | *** join/#asterisk |Vulture| (Vulture@dhcp64-134-129-79.wbt.phx.wayport.net) |
| 05:17.52 | Essobi | How much is a 2600 with a T1 wic new? |
| 05:17.53 | SexyKen | That doesn't seem to be working. |
| 05:18.00 | shepherd | agi: what's wrong with that? |
| 05:18.03 | shepherd | :D |
| 05:18.03 | opus___ | yeah |
| 05:18.05 | SexyKen | It doesn't log the initial incoming. |
| 05:18.06 | AgiNamu | haha |
| 05:18.11 | opus___ | ajsdlfka |
| 05:18.15 | Essobi | ymmm |
| 05:18.17 | AgiNamu | yea, just like ordering new whatevers to ensure you use your budge :P |
| 05:18.20 | |Vulture| | If you have a PRI and you just have 1 DID, when it rings in it goes to the first available channel, correct? |
| 05:18.28 | Mikrozaur | Anybody speak in Russian ? |
| 05:18.33 | PTG123 | SexyKen: you alive? |
| 05:18.34 | alexns | eight check my dcc |
| 05:18.36 | AgiNamu | Mikrozaur, probably many people. |
| 05:18.42 | Essobi | vult usually. |
| 05:18.42 | SexyKen | Pres, Yea, I'm here. |
| 05:18.43 | puppet | mikrozaur: but not here maybe |
| 05:18.47 | Eight | alexns: ya, that's about what I"m using. But I got it working =) Just not with the chi' proxy =/ |
| 05:18.51 | PTG123 | SexyKen: heh im? :) |
| 05:18.54 | |Vulture| | Essobi: there are exceptions? |
| 05:18.56 | shepherd | mikrozaur: matt fredrickson @ digium does :) |
| 05:19.00 | alexns | sorry |
| 05:19.08 | Essobi | I think it depends on the protocol on the PRI. |
| 05:19.09 | AgiNamu | hmm... vodka! |
| 05:19.13 | AgiNamu | That's all the russian i know. |
| 05:19.14 | alexns | does chi use old authentication |
| 05:19.20 | `Sauron | Most PRI's will ring in the first available channel |
| 05:19.22 | shepherd | privot! |
| 05:19.23 | alexns | i get bad invite |
| 05:19.25 | |Vulture| | Essobi: okay |
| 05:19.26 | `Sauron | if you dail the DID several times |
| 05:19.35 | Mikrozaur | AgiNamu: It's vreri good to. But I |
| 05:19.42 | Essobi | 99% ring first available from the top, I believe thou. |
| 05:19.55 | Mikrozaur | AgiNamu: It's vreri good to. But I'm have momeone questions about asterisk |
| 05:20.30 | Essobi | Anyone looked at this? http://www.soekris.com/bundles.htm |
| 05:20.31 | AgiNamu | well, try in english |
| 05:20.55 | Essobi | I can't remember how much an 2600 w/T1 costs thou.. |
| 05:20.57 | *** join/#asterisk zignig (~simon@203.217.15.10) |
| 05:21.15 | opus___ | essobi - have an older version at the office |
| 05:21.30 | Essobi | $646 seems kind of high for a single T1 port thou. |
| 05:21.31 | opus___ | extremely low power consumption, amd 586 chip |
| 05:21.42 | Essobi | Are they a POS? |
| 05:21.51 | Mikrozaur | I'm have asterisk server on linux 2.6 and xlite soft dilaer for testing it. Dillino fron one PC to annower its ok. But I'm not can strart confirence room. app_meetme started. |
| 05:22.09 | opus___ | nope, pretty well engineered. I don't have the T1 version |
| 05:22.11 | Essobi | They are bundling them with a Sangoma E1/T1 card.. |
| 05:22.16 | Mikrozaur | exten => 1000,1,MeetMe(2222) |
| 05:22.25 | shepherd | essboi: boooo sangoma |
| 05:22.28 | Essobi | I'm wondering if the CPU can atually drive a Full T1.. |
| 05:22.29 | puppet | mikrozaur: for meetme you need to have zaptel compiled u got that? |
| 05:22.37 | AgiNamu | Sweet shit! I'm getting a RJ11->USB FireFly-compatible ATA |
| 05:22.40 | Mikrozaur | conf => 2222 |
| 05:23.19 | Mikrozaur | Yes im' alredi compiled it. If not app_meetme not compiling. |
| 05:23.27 | puppet | mikrozaur: do du -s /usr/lib/asterisk/modules/app_meetme.so |
| 05:24.02 | puppet | "du -s /usr/lib/asterisk/modules/app_meetme.so" |
| 05:24.04 | puppet | that is |
| 05:24.29 | Mikrozaur | But if i'm call to 1000 on console i'm see playng(conf-invalid) |
| 05:24.43 | Mikrozaur | 96 /usr/lib/asterisk/modules/app_meetme.so |
| 05:24.50 | puppet | ok :) |
| 05:25.02 | *** join/#asterisk paulc (paulc@176.134.218.209.transedge.com) |
| 05:25.21 | Mikrozaur | What i'm doing not right ? |
| 05:25.39 | puppet | something :) |
| 05:25.45 | Eight | Mikrozaur: you need ZTDummy |
| 05:25.57 | Eight | Mikrozaur: the conference room doesn't work without ZTDummy *or* zaptel hardware. |
| 05:26.09 | Eight | Mikrozaur: the wiki has info on it. |
| 05:26.13 | Eight | jbot ztdummy? |
| 05:26.14 | jbot | somebody said ztdummy was zaptel timing source which uses a usb-ohci compatible usb controller as source. (part of zaptel cvs) |
| 05:26.26 | Eight | how do I add info to jbot? |
| 05:26.31 | Essobi | net4801-50: 266 Mhz CPU, 128 Mbyte SDRAM, 3 Ethernet, 2 serial, USB connector, CF socket, 44 pins IDE connector, 1 Mini-PCI socket, 3.3V PCI connector. .... Plus a T1 card in ... |
| 05:26.46 | `Sauron | ztdummy works great |
| 05:26.49 | Essobi | Is there zaptel drivers for FreeBSD yet? |
| 05:26.52 | opus___ | whoah, whats a good price for a voice T1 |
| 05:26.55 | Mikrozaur | Whot is ZTDummy ? I'm colpile zaptel. But i'm not have hardware |
| 05:26.55 | `Sauron | if you're on 2.6 you don't need the USB timer |
| 05:27.05 | puppet | mikrozaur: since you are russing, is there any providers for free russian number for forwarding to a sip or just a sipprovider? |
| 05:27.40 | opus___ | essobi - it must do the codec in the fpga |
| 05:28.11 | *** join/#asterisk kks (~kks@203.115.210.253) |
| 05:28.24 | Mikrozaur | puppet:No i'm not from russian. i'm from uzbekistan/ Whre not VoIP providers |
| 05:28.31 | puppet | mikrozaur: ok |
| 05:29.45 | *** join/#asterisk mrgoby (~mrgoby@141.211.162.97) |
| 05:30.05 | mrgoby | i'm having problems with Redirection from the managment interface |
| 05:30.27 | puppet | junk-c: one more ;p |
| 05:30.50 | mrgoby | <PROTECTED> |
| 05:31.20 | Essobi | Oh fuck me.. There's FreeBSD drivers for Sangoma T1 cards.. Is it possible? We can have full fledged FreeBSD * routers finally? |
| 05:31.26 | Eight | I had to go and change my phone number, now BV is broken again =p |
| 05:31.46 | puppet | essobi: girl rl? then yes else ill skip it :) |
| 05:31.58 | Essobi | puppet Go away, sock boy. :) |
| 05:32.03 | puppet | haha |
| 05:32.23 | puppet | < compiling lame and sox on new now so i get mp3 support so i dont have to mail 1mb voicebox :) |
| 05:32.33 | Essobi | MAhaha |
| 05:32.43 | shepherd | essobi: just because their are drives doesn't mean it will work with asterisk |
| 05:32.46 | Mikrozaur | <PROTECTED> |
| 05:32.46 | Mikrozaur | <PROTECTED> |
| 05:32.48 | Mikrozaur | <PROTECTED> |
| 05:32.48 | Mikrozaur | <PROTECTED> |
| 05:32.48 | shepherd | there |
| 05:33.03 | puppet | mikrozaur: please use pastebin.ca for much text :) |
| 05:33.38 | Mikrozaur | ? |
| 05:33.38 | *** join/#asterisk MicH323 (~micosat@host-84-9-63-27.bulldogdsl.com) |
| 05:34.01 | shepherd | http://pastebin.ca |
| 05:34.27 | Mikrozaur | i' see |
| 05:35.11 | puppet | opus___: did u get it to work? |
| 05:35.19 | Mikrozaur | It's http://pastebin.ca/raw/7223 ? |
| 05:35.28 | opus___ | <PROTECTED> |
| 05:35.32 | puppet | opus___: sweet |
| 05:35.38 | puppet | opus___: im gone reconfigure mine soon :) |
| 05:35.50 | puppet | opus___: i miss feature to be able to pick calls in queue tho :p |
| 05:35.50 | Funbags | Eight, whats wrong w/bv tonight? |
| 05:35.56 | opus___ | a killall -9 perl save the day |
| 05:35.59 | puppet | opus___: so if 5 people are in queue i pick out one |
| 05:36.12 | puppet | opus___: not auto, that would rock ;p |
| 05:37.22 | opus___ | check this out |
| 05:37.41 | puppet | opus___: http://sip.puppet.nu/sip/ my phat n00b setup ;p |
| 05:38.09 | opus___ | http://pastebin.ca/7224 |
| 05:38.12 | Eight | Funbags: the chi' proxy doesn't have my account info, and it broke when I changed my phone numbers =p |
| 05:38.14 | Eight | Funbags: other than that =) |
| 05:38.23 | opus___ | Save that to a file called "manager.hta" and in windows it will make it an application |
| 05:38.27 | Funbags | Eight, ahh |
| 05:38.38 | opus___ | once you click on it |
| 05:38.57 | alexns | pupet: that shit is decent |
| 05:39.01 | Eight | Funbags: right now I'm recompiling asterisk without a certain patch... it didn't fix my original problem with the chi' proxy. I want to see if it was necessary at all. |
| 05:39.03 | mrgoby | anyone have redirects working in php ? |
| 05:39.11 | alexns | redirect? |
| 05:39.16 | mrgoby | management intface redirect |
| 05:39.16 | alexns | sure |
| 05:39.18 | Eight | I want to be able to add on the Wiki that 1.0.6 works with BV... if it does. |
| 05:39.19 | puppet | opus___: cool ;D |
| 05:39.24 | Funbags | Eight, the callerid when somone calls you pass thru asterisk ok to your sip devices?? |
| 05:39.34 | Funbags | Eight, w/bv |
| 05:39.36 | mrgoby | can i see a code snippet ? i have no idea why this isnt working |
| 05:39.37 | mrgoby | ? |
| 05:39.40 | alexns | hehe i only know how to dor url redirect |
| 05:39.45 | alexns | in php |
| 05:39.50 | Eight | Funbags: I was about to check out callerID stuff, when I changed numbers and broke everything =p |
| 05:39.52 | mrgoby | oh |
| 05:39.59 | puppet | mrgoby: should work yes |
| 05:40.12 | mrgoby | should is operative |
| 05:40.13 | puppet | mrgoby: just use same syntax as you would do in any other place |
| 05:40.18 | Funbags | Eight, when someone calls me for some reason i dont see caller id |
| 05:40.25 | Funbags | Eight, yet i see it in the debug |
| 05:40.38 | mrgoby | i get this back from the management intface |
| 05:40.48 | mrgoby | Response: Success |
| 05:40.50 | *** join/#asterisk roamer323 (~sing@70.50.113.2) |
| 05:40.56 | mrgoby | Message: Redirect successful |
| 05:41.01 | mrgoby | but it doesnt work |
| 05:41.20 | Eight | oooh duh. |
| 05:41.23 | mrgoby | do redirects work if you are not currently using an app such as dial ? |
| 05:41.23 | *** join/#asterisk Tarox (someone@pD9E7B6B5.dip.t-dialin.net) |
| 05:41.23 | puppet | Supported file formats: aiff al alsa au auto avr cdr cvs dat vms gsm hcom la lu maud mp3 nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw ub ul uw voc vorbis vox wav wve |
| 05:41.25 | Eight | THAT explains why BV broke... |
| 05:41.27 | puppet | yeah! ;D |
| 05:41.36 | Eight | my accountID is my phone number, if I change my phone number I need to change my accountID stuff. |
| 05:41.56 | Eight | This is why ID columns are ARBITRARY =p |
| 05:42.02 | mrgoby | the people i'm trying to redirect are sitting on a background |
| 05:42.13 | mrgoby | and i'm trying to push them to a meetme |
| 05:42.25 | opus___ | cool |
| 05:42.33 | mrgoby | but nothing works |
| 05:46.39 | Eight | lol, nice. |
| 05:46.52 | Eight | now my AuthID is my old phone number, but I have to use my new phone number in the registration. |
| 05:46.57 | puppet | hmmpf |
| 05:47.05 | Eight | And the old phone number is disconnected... |
| 05:47.08 | Eight | that's cute =p |
| 05:47.23 | puppet | Can I just do MONITOR_EXEC=somepath anywhere i want? for a default one? |
| 05:47.40 | *** join/#asterisk file (~file@251.134.218.209.transedge.com) |
| 05:49.11 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 05:49.35 | *** join/#asterisk lqdengr (~fweston@69.172.205.68.cfl.res.rr.com) |
| 05:51.05 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 05:52.23 | hardwire | boop boop de doop boop |
| 05:53.12 | Nukemizer | Does anyone use fax receive with a digium card that would be willing to guide me on a dialplan for fax detection on an incoming call ? |
| 05:54.01 | Nukemizer | or perhaps I should ask.. can I not get fax detection if I have call answered by a main menu and greeting ? |
| 05:54.26 | *** join/#asterisk amsterdam (~ak@p5087B56A.dip.t-dialin.net) |
| 05:55.31 | puppet | woho |
| 05:55.39 | hardwire | osp? |
| 05:55.46 | puppet | outgoing-073xxxxxxx_20050311-065128.mp3 |
| 05:55.48 | puppet | ;D |
| 05:57.06 | hardwire | jesus |
| 05:57.12 | hardwire | app_rpt exists! |
| 05:57.30 | *** join/#asterisk |vorlon (~vorlon@251.134.218.209.transedge.com) |
| 05:57.32 | mrgoby | interesting.... it is transfering me to the context i want, but not the extension |
| 05:57.47 | puppet | mrgoby: i got same problem |
| 05:58.20 | puppet | mrgoby: im trying to transfer calls to f.ex a mailbox but i just get to context seems like it always goes to s, |
| 05:58.56 | mrgoby | it doesnt even go to s in my case |
| 05:58.59 | mrgoby | just hangs |
| 05:59.11 | mrgoby | i specified s even |
| 05:59.20 | mrgoby | to try to workaround |
| 05:59.38 | puppet | hmm works "fine" in op_panel except it goes to s, and not the extension i want |
| 06:00.11 | puppet | hm gone try a thing brb |
| 06:02.07 | *** join/#asterisk ikey (~kirankuma@202.54.37.186) |
| 06:02.15 | *** join/#asterisk channan (~channan9@66.180.121.185) |
| 06:05.22 | puppet | woot it works now |
| 06:05.28 | puppet | mrgoby: thanks ;D i had totally forgot that |
| 06:05.35 | puppet | nowi can connect calls to my voicebox ;p |
| 06:05.37 | file | oe |
| 06:05.48 | puppet | wich i just could connect them to a queue where i could pick them up again |
| 06:06.51 | mrgoby | i cant figure out where the heck it is transfering my call to |
| 06:07.35 | puppet | mrgoby: hold a sec and i can maybe explain some more :) |
| 06:08.26 | mrgoby | ok, pleeeez do :-D |
| 06:08.33 | mrgoby | this is driving me crazy |
| 06:09.08 | Eight | anyone here familiar with using a DNS SRV record for SIP direction? |
| 06:09.08 | puppet | seems to be trouble to listen to commands after it have been redirected some hmm *tests more* |
| 06:10.03 | puppet | nah works :) |
| 06:10.12 | *** join/#asterisk clive- (~pirch@rndf-146-55-139.telkomadsl.co.za) |
| 06:10.58 | Eight | My understanding is if someone tries to dial myentry@mydomain.net in their sip client, and I have a "_sip_udp.mydomain.net. IN SRV 20 0 5060 sipserver.mydomain.net." in my DNS record, they'll try to connect to myentry on sipserver.mydomain.net ? |
| 06:11.33 | puppet | is it true?! ;D |
| 06:11.35 | puppet | thats cool |
| 06:11.49 | BBRodriguez | has anyone tried www.voipescrow2.net with asterisk ? |
| 06:11.55 | Eight | puppet: it does seem cool. |
| 06:12.03 | Eight | It's just the equivalent of an MX record for mail, which is swell. |
| 06:12.08 | puppet | eight: ill check around later today |
| 06:12.21 | puppet | eight: right now i want to get other stuff up running |
| 06:12.22 | Eight | puppet: Cool, I'm still digging for info on it... |
| 06:12.29 | puppet | f.ex working voicemail |
| 06:12.33 | xeet2 | what does a call on a pri show up as in *? Zap/1-num ? (span = 1) |
| 06:12.48 | puppet | mrgoby: check msgs :) |
| 06:15.25 | Eight | I think I might have to do something special with the authentication though... |
| 06:16.22 | Eight | "With DNS SRV records you bind a SIP proxy to a specific domain, to enable URL calling to domains, instead of addressing a specific SIP proxy." |
| 06:16.31 | Eight | Ya, that's totally 'the shit'. |
| 06:16.40 | opus___ | what does the "_" character do under a dial plan for extensions.conf? |
| 06:16.44 | opus___ | is it *? |
| 06:16.47 | Eight | opus___: no... |
| 06:16.57 | Eight | opus___: it tells it to read the extension as a pattern, instead of literally. |
| 06:17.13 | opus___ | oh |
| 06:17.21 | Eight | you can use stuff like ., N, X, etc with you preceed with _ |
| 06:17.32 | opus___ | reminds me of MCI codes on old bbs systems |
| 06:17.40 | Eight | er, precede. |
| 06:18.06 | Eight | hmm... I think I'm actually answering more questions than I'm asking in this channel =) |
| 06:18.09 | Eight | Go me =) |
| 06:18.22 | xeet2 | hehe |
| 06:22.14 | zignig | Eight: yay for you. |
| 06:22.29 | zignig | means that you _may_ even know what you are talking about ..... ;) |
| 06:22.39 | Eight | zignig: ahaha... no chance. |
| 06:22.50 | zignig | :) |
| 06:25.08 | zignig | has anyone got db based extension routing going ? |
| 06:25.20 | zignig | ie call comes in , check against database |
| 06:25.35 | zignig | route to phone , blacklist , whitelist , message bank etc ? |
| 06:25.42 | Eight | jbot mysql? |
| 06:25.43 | jbot | SQL (Structured Query Language) database server. URL: http://www.mysql.com/ |
| 06:26.57 | *** join/#asterisk tuxinator_linuxM (~tuxinator@251.134.218.209.transedge.com) |
| 06:30.22 | *** join/#asterisk three55ml (~who@cpe-66-68-110-140.austin.res.rr.com) |
| 06:30.36 | three55ml | Anyone use a DVG-1120? I'm having some incoming caller ID issues. |
| 06:31.48 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:33.33 | kks | Anyone use Festival(method 1), i get UDP checksum error. Any recommendation? |
| 06:33.39 | opus___ | damn you broadvoice |
| 06:33.41 | *** join/#asterisk datareactor (~datareact@203.81.192.53) |
| 06:33.44 | Eight | opus___: heheh. |
| 06:33.51 | puppet | eight: anything new about it? |
| 06:33.55 | puppet | kks: run it in AGI |
| 06:33.58 | Eight | puppet: SRV? nope. |
| 06:34.01 | puppet | kks: search wiki for how to do it |
| 06:34.04 | puppet | eight: ok :) |
| 06:34.27 | puppet | kks: want example how endresult gets when using it in AGI msg me for number to call |
| 06:34.59 | puppet | eight: but how can i call with @ in xlite? ;P u cant? ;P |
| 06:35.05 | kks | puppet, i try that also...but no voice wiht perl script |
| 06:35.14 | puppet | kks then youve done wrong :) |
| 06:36.15 | Eight | puppet: should be able to in SJPhone. |
| 06:37.35 | mog_home | twisted i like the time stap app |
| 06:37.38 | mog_home | stamp* |
| 06:37.40 | mog_home | very cool |
| 06:37.56 | kks | puppet, when using agi i get the error msg with "not exit file format" then i modify it with the full path |
| 06:38.17 | SexyKen | Anyone know how I can set the userfield on a record and then have it passed to each additional record created on that same call? |
| 06:38.28 | kks | puppet, however cmd Festival() produce very bad and unclear voice |
| 06:38.42 | Damin | Anyone interested in helping out, there is a general bug-tracker squashing session going on in #asterisk-bugs, w/ many of the developers who are at VON. We're trying to get the Bug Tracker cleaned up a bit.. |
| 06:39.19 | Juggie | most of the bugs on the tracker require extensive work |
| 06:40.06 | SexyKen | Juggie, is there a costco near you? |
| 06:40.08 | *** join/#asterisk finejava (~abc@218.208.119.98) |
| 06:40.19 | Damin | Juggie: That's why we're working on smaller, quicker patches tonight to clean out the cruft and get focused for more extensive work.. |
| 06:40.24 | Eight | Man, is it just my poor luck or is the Asterisk community prone to shouting matches? (users list, and this channel to an extent) |
| 06:40.34 | Juggie | SexyKen, yes, why |
| 06:40.42 | finejava | ermm...hi guys |
| 06:40.46 | Nugget | WE NEVER SHOUT! |
| 06:40.46 | finejava | i'm new here |
| 06:40.53 | Eight | finejava: welcome. |
| 06:40.56 | finejava | can i know who is the digium marshall |
| 06:41.07 | SexyKen | Juggie, run down there and get a box of RAID. Then just kill all of the bugs. |
| 06:41.19 | finejava | i've got an issue with asterisk that i need to talk about |
| 06:41.24 | Juggie | SexyKen, i forgot to laugh. |
| 06:41.37 | SexyKen | finejava, Go to #counseling -- they'll help you with your issue. |
| 06:41.55 | finejava | thx sexyken |
| 06:42.06 | SexyKen | finejava....I'm not serious. |
| 06:42.20 | finejava | ehmm... |
| 06:42.33 | finejava | anyway... |
| 06:42.41 | finejava | who should i refer too... |
| 06:42.42 | Eight | http://lists.digium.com/pipermail/asterisk-users/2004-June/049190.html |
| 06:42.50 | hardwire | anybody know how to fix the scrollback buffer issues with the cli |
| 06:42.50 | SexyKen | Refer to the channel. |
| 06:42.51 | hardwire | or rather the command history.. sorry |
| 06:42.53 | Eight | looks like a decent "what is DNS SRV, re: SIP" |
| 06:42.53 | hardwire | *CLI> stop\040now |
| 06:42.55 | hardwire | is that just a terminal issue? |
| 06:43.10 | Juggie | hardwire, yes |
| 06:43.14 | Juggie | what terminal are you using |
| 06:43.22 | Eight | Seems like the only contention about DNS SRV w/SIP is by people who have no idea how e-mail works =p |
| 06:43.37 | hardwire | Juggie: linux/vt100/xterm |
| 06:43.40 | hardwire | none seem to work :) |
| 06:43.51 | finejava | ehmm...looks like everyone is talking their probolems here... |
| 06:43.58 | hardwire | should I change from en_us to C ? |
| 06:44.09 | Juggie | hardwire, i've never seen any such problem |
| 06:44.14 | finejava | i've got an issue for one way voice or both |
| 06:44.52 | finejava | the thing is...asterisk...is still sending invite SDP even though the call has been bridge |
| 06:44.59 | hardwire | Juggie: what terminal or shell are you using? |
| 06:45.01 | finejava | and 2 of my box are behind a nat |
| 06:45.18 | Juggie | hardwire, putty,securecrt with their default settings |
| 06:45.26 | Juggie | and also a local term |
| 06:45.40 | hardwire | bash? |
| 06:45.47 | hardwire | default profiles to what OS? |
| 06:46.18 | finejava | the questio is...y is asterisk still sending invites even though the calls have been bridge |
| 06:46.28 | finejava | can anyone help??? |
| 06:46.34 | kram | because it's trying to native bridge |
| 06:46.36 | kram | unless you turn it off |
| 06:46.39 | hardwire | Juggie: heh.. can you send me a few safe lines from your .asterisk_history? |
| 06:47.06 | finejava | the thing is...we dun wan the calls rtp to go thru the proxy |
| 06:47.10 | hardwire | sip\134134134134134134134134134134134134040debug\134134134134134134134134134134134012\134134134134134134134134134134012\134134134134134134134134$stop\134134134134134134134134134134134134040now\134134134134134134134134134134134012\134134134134134134134134134134012\1341341341341341341341341$ |
| 06:47.13 | finejava | it will choke the server |
| 06:47.13 | hardwire | theres a few lines from mine |
| 06:47.28 | Juggie | hardwire, mine has nothing of the sort, just plain text commands seperated by a blank line |
| 06:47.43 | hardwire | hehe |
| 06:47.45 | hardwire | well lucky you |
| 06:47.45 | Eight | I thought the only reason Ast' would send extra invites after the call is going is to try to get the two callers to 'peer' directly, unless either sip.conf entry has canreinvite=no, OR the call was made with T or t options. |
| 06:47.53 | finejava | therefore we set the canreinvite to yes |
| 06:48.30 | Eight | finejava: How many calls are you talking? |
| 06:48.31 | finejava | currenly...my dialing stirng is sip/${EXTEN}|60 |
| 06:48.53 | finejava | we r running a production server... |
| 06:49.21 | Eight | finejava: Ok. I"m not quite sure what your problem is exactly? |
| 06:49.33 | Eight | finejava: You *want* Ast' to tell teh callers to peer directly? |
| 06:49.35 | Eight | finejava: but it's not? |
| 06:49.36 | finejava | ok... |
| 06:49.47 | finejava | i'll explain in details... |
| 06:49.53 | mrgoby | anyone have funny experience with zaptel and redirects ? |
| 06:50.00 | finejava | here is the scenario's |
| 06:50.09 | mrgoby | it is redirecting me to the priority i specify minus 1 |
| 06:50.12 | finejava | i've got 1 box behind a NAT |
| 06:50.22 | PTG123 | any sql people awake? :) |
| 06:50.22 | Eight | ah =) |
| 06:50.25 | finejava | and another box behind another NAT |
| 06:50.37 | mrgoby | and doesnt actually execute that extension's app, though show channels shows me on that extension and priority |
| 06:50.39 | Eight | finejava: good luck with that, then. |
| 06:50.45 | PTG123 | mysql> select COUNT(calldate) AS COUNT,channel,SUM(duration) AS DURATION FROM cdr WHERE channel LIKE "IAX2%"; |
| 06:50.45 | PTG123 | ERROR 1140: Mixing of GROUP columns (MIN(),MAX(),COUNT()...) with no GROUP columns is illegal if there is no GROUP BY clause |
| 06:50.45 | finejava | and this 2 box are on different part of the world |
| 06:50.50 | PTG123 | anyone see a prob with that query? |
| 06:50.53 | mrgoby | very strange stuff.... |
| 06:51.01 | Eight | finejava: It's possible to get that to work, I think. But it's a giant headache, afaik. |
| 06:51.14 | Eight | finejava: You're going to, most likely, just have to tunnel through ast' as normal. |
| 06:51.17 | finejava | the problem occurs with certain routers |
| 06:51.31 | finejava | like dlink |
| 06:51.42 | finejava | when a party calls... |
| 06:51.51 | finejava | asterisk send an invite |
| 06:52.06 | Eight | ok, layout for me where everyone is? |
| 06:52.26 | Eight | Client->Ast->NAT->internet->NAT->Client? |
| 06:52.35 | finejava | yeah |
| 06:52.49 | finejava | the problem is... |
| 06:52.49 | Eight | And you're getting the ring, but no audio? |
| 06:52.56 | finejava | yeah |
| 06:53.05 | finejava | we got a team... |
| 06:53.18 | finejava | and we found the problem |
| 06:53.26 | Eight | The problem is Ast sees ClientA (the caller) as, say, 192.168.1.something. |
| 06:53.29 | finejava | 1st invite sends the local IP |
| 06:53.35 | Qwell | PTG123: max(channel) |
| 06:53.37 | finejava | yeah u r rite |
| 06:53.42 | Qwell | SQL is stupid... |
| 06:53.52 | Eight | So when Ast' tries to tell ClientB to talk directly to ClientA, it says go talk to 192.168.1.something |
| 06:53.58 | Eight | Which obviously isn't the right place. |
| 06:54.01 | xeet2 | what is the range for good test results from zttest? |
| 06:54.03 | tuxinator_linuxM | Qwell: I don' t think you SQL that well |
| 06:54.13 | Eight | what is clientA running? X-lite? |
| 06:54.22 | Eight | If it's X-lite you can set 'use internal IP: NEVER' |
| 06:54.27 | Qwell | tuxinator_linuxM: I just know that when I write SQL between MySQL and SQL server, it makes me do retarded things like that. |
| 06:54.33 | Juggie | finejava, there is a way to set local and external ip's in your sip.conf |
| 06:54.37 | finejava | clientA voip adapter...clientB voip adapter |
| 06:54.38 | Qwell | or, I have to put ALL of the fields into a group by...which takes more time. |
| 06:54.40 | Juggie | check the wiki for more information |
| 06:54.50 | Eight | finejava: I think you're screwed =/ |
| 06:54.58 | Juggie | no, hes not |
| 06:55.12 | Qwell | either way will work, and return the same results |
| 06:55.16 | Eight | Juggie: that's for Ast', but not the clients. |
| 06:55.19 | Eight | Juggie: right? |
| 06:55.25 | finejava | the thing is... |
| 06:55.37 | finejava | if asterisk stop sending invites after the 3rd invites |
| 06:55.39 | finejava | it works... |
| 06:55.46 | Qwell | and, hmm, that is mysql...weird, never had to write ugly sql with mysql |
| 06:55.54 | finejava | the 4th invites causes the whole thing not to work... |
| 06:55.59 | Juggie | yes eight... |
| 06:56.01 | finejava | because the 4th invites send the local IP... |
| 06:56.07 | Nugget | that's because mysql only supports a ridiculously small subset of sql. |
| 06:56.13 | Juggie | it still should not be a huge issue |
| 06:56.16 | Juggie | xten can deal with it |
| 06:56.25 | Eight | Juggie: he's not using Xten, though. |
| 06:56.26 | Nugget | it doesn't get ugly until you are able to actually do interesting things |
| 06:56.28 | Qwell | maybe I just needed to finish reading the error... |
| 06:56.34 | Juggie | what is it, a hard phone? |
| 06:56.36 | Eight | finejava: Sipura 1001 or something? |
| 06:56.42 | opus___ | uh, |
| 06:56.44 | finejava | yeah hard phone |
| 06:56.44 | opus___ | Background info: It's not the card that detects the fax. Its the builtin code in asterisk that does it (dsp.c). chan_zap.c is currently the only channel driver that uses the faxdetection but in theory it could be enabled/used in other channel drivers as well. |
| 06:56.48 | Juggie | which one |
| 06:56.48 | finejava | we try on sipura |
| 06:56.51 | finejava | linksys |
| 06:57.04 | opus___ | that'd be awesome to get a fax over broadvoice on the mainline |
| 06:57.10 | Eight | finejava: You have to get ClientA to identify itself as the public IP, afaik. |
| 06:57.13 | Juggie | finejava, the problem is when your client registers to * it reports its ip as the internal ip |
| 06:57.15 | finejava | ours is similar to sipura |
| 06:57.19 | *** join/#asterisk viLeR (1000@ip-47-233.telesat.com.co) |
| 06:57.26 | xeet2 | opus: agreed =P |
| 06:57.46 | finejava | the thing is...in asterisk...i can c the wan ip and local ip |
| 06:58.07 | Juggie | finejava, did you set nat=yes for the client? |
| 06:58.15 | Juggie | i'm not sure how * would handle this. |
| 06:58.16 | finejava | yeah |
| 06:58.19 | finejava | i've got nat=yes |
| 06:58.20 | Nugget | finejava: as asterisk behind nat or is it just multi-homed? |
| 06:58.29 | Juggie | now, you also know that there needs to be a route for return traffic |
| 06:58.30 | finejava | asterisk is not behind nat |
| 06:58.38 | Nugget | that's how mine is, it works just fine. |
| 06:58.40 | finejava | i've also got canreinvite=yes |
| 06:58.40 | Juggie | but your phones are. |
| 06:58.40 | Eight | Ok, well, I'm out of ideas. I'm just going to listen now, because I'm curious how this can be solved myself. |
| 06:58.45 | Juggie | doesnt matter.... |
| 06:58.52 | Juggie | say i am trying to send your phone information |
| 06:58.53 | Juggie | its udp |
| 06:58.53 | Eight | Nugget: but you're not doing reinvites (eg, having the clients peer) right? |
| 06:58.54 | finejava | basically |
| 06:58.58 | Juggie | it needs to get into your phone |
| 06:59.00 | Juggie | through the nat |
| 06:59.08 | finejava | we have the solution |
| 06:59.12 | Nugget | there's no nat anywhere in my setup. not for asterisk or for the devices. |
| 06:59.18 | Juggie | which is |
| 06:59.26 | Nugget | the only complexity is that the asterisk box sits on private ip space and public ip space |
| 06:59.27 | finejava | the solution is to remove the 4th invites |
| 06:59.30 | Eight | Nugget: ah, so the ClientA is on a public IP. That works too =) |
| 06:59.38 | Nugget | no, the clients are all on private ip space |
| 06:59.40 | finejava | because the 4th invites give's the lan ip |
| 06:59.47 | Juggie | finejava huh? |
| 06:59.52 | Nugget | they can't even route to the internet |
| 07:00.03 | Eight | Nugget: Ah. |
| 07:00.18 | Juggie | finejava, you cant support more then one phone on the other side of this nat either really |
| 07:00.22 | Juggie | because the packets have no way in |
| 07:00.44 | finejava | the packets have the way in |
| 07:00.52 | finejava | just that we need to eliminate the 4th invites |
| 07:01.08 | finejava | we've got another scenario... |
| 07:01.18 | finejava | cleintA dials clientB |
| 07:01.25 | finejava | clientB pick up |
| 07:01.41 | finejava | both party can hear each other for 1 sec... |
| 07:01.47 | finejava | then no more voice |
| 07:02.01 | Juggie | yes that is definitally nat |
| 07:02.01 | finejava | this is due to...asterisk subsequently...sending invites to the box |
| 07:02.30 | finejava | where the invites contact is lan ip |
| 07:02.52 | finejava | there for...the voip adapter...accept the invites...and gone the voice... |
| 07:02.52 | *** join/#asterisk DaLion (DaLion@Toronto-HSE-ppp3770824.sympatico.ca) |
| 07:03.10 | *** join/#asterisk mog_home (~mog_home@146.229.176.173) |
| 07:03.27 | finejava | thr rtp packet routers to the lan ip... |
| 07:03.55 | finejava | i've got a guy who is writing the firmware for our adapter... |
| 07:04.11 | DaLion | who uses sveasoft and what diff betwene satoi and alcemy |
| 07:04.35 | finejava | he said that...if we can eliminate the 4th invites...or...ask asterisk to sto sending invites after both party are connected than we r done... |
| 07:04.59 | finejava | cuase the culprit is the 4th invites which sends the lan ip... |
| 07:05.03 | shepherd | http://www.voip-news.com/art/3y.html |
| 07:05.05 | shepherd | scary stuff |
| 07:05.16 | Juggie | finejava, then submit a bug and patch to the bug tracker |
| 07:05.23 | finejava | i've submit... |
| 07:05.33 | Juggie | did you submit the patch? |
| 07:05.37 | finejava | mark close it...cause he say it's configuration issue |
| 07:05.41 | finejava | which it's not... |
| 07:06.19 | finejava | and he ask me to talk to the marshall...b4 reopening the bug |
| 07:06.26 | finejava | my bug id is 3750 |
| 07:06.32 | Juggie | then prove that its not |
| 07:06.39 | Juggie | attach a sip debug log |
| 07:06.45 | Juggie | document it to where it dos the 4th invite |
| 07:06.48 | Juggie | *does |
| 07:07.13 | finejava | u can goto the website |
| 07:07.13 | shepherd | heh |
| 07:07.17 | finejava | i've attach the trace log |
| 07:07.21 | shepherd | if it's sip, mark probably doesn't want to fix it |
| 07:07.23 | shepherd | haha |
| 07:07.25 | finejava | ethereal.dat |
| 07:08.27 | *** join/#asterisk Inv_arp (junya@adsl-8-232-168.mia.bellsouth.net) |
| 07:09.41 | Juggie | finejava, does your ast have an internet ip |
| 07:09.44 | Juggie | or is it local |
| 07:10.01 | tuxinator_linuxM | ~seen ciesl1n |
| 07:10.06 | jbot | i haven't seen 'ciesl1n', tuxinator_linuxM |
| 07:12.36 | SexyKen | Has anyone here ever heard of Bicom Systems? |
| 07:13.15 | *** join/#asterisk grandjava (~abc@218.208.119.98) |
| 07:13.21 | grandjava | hi sorry i was dc |
| 07:13.25 | grandjava | i'm finejava |
| 07:13.38 | grandjava | did i miss any msg |
| 07:15.45 | grandjava | is there any configuration in my sip.conf |
| 07:15.51 | grandjava | or...it's a bug |
| 07:17.46 | grandjava | hello??? |
| 07:17.51 | *** part/#asterisk grandjava (~abc@218.208.119.98) |
| 07:17.57 | *** join/#asterisk grandjava (~abc@218.208.119.98) |
| 07:18.06 | grandjava | sorry dc again |
| 07:18.12 | grandjava | finejava here |
| 07:18.27 | grandjava | darn my broadband is pissing me |
| 07:18.35 | opus___ | does spandsp work with cvs head? |
| 07:18.41 | opus___ | or 1.0.6 for that matter.. |
| 07:19.05 | grandjava | r u there eight/juggie??? |
| 07:19.43 | MicH323 | . |
| 07:19.49 | Nugget | i m hr 2! |
| 07:20.03 | Nugget | r u 2 a nw alo bi chn? |
| 07:20.26 | grandjava | ??? |
| 07:20.36 | Nugget | !!! |
| 07:20.41 | grandjava | is there anyway to solve the problem i'm having... |
| 07:21.05 | MicH323 | whats your prob? |
| 07:21.12 | grandjava | ermm... |
| 07:21.17 | *** join/#asterisk Jas_Williams (~jas_willi@217.41.229.134) |
| 07:21.23 | MicH323 | soory.. finejava!!! |
| 07:21.38 | grandjava | the 4th invites is causing the calls not to go thru |
| 07:21.59 | grandjava | because the 4th invites is sending the lan IP instead of the wan IP |
| 07:22.18 | grandjava | so i need to elimnate the 4th invites for the calls to be successful... |
| 07:22.32 | *** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com) |
| 07:22.36 | grandjava | and * is keep on sending invites even thoght the calls is bridged... |
| 07:22.43 | MicH323 | I tried with Cisco ATA and it seems to work. I thought the Linksys would be the same |
| 07:23.13 | grandjava | the adapter will need to be behind a NAT |
| 07:23.21 | grandjava | it only happens with certain router |
| 07:23.37 | grandjava | one thing for sure...dlink is one of them |
| 07:24.53 | grandjava | is there anyway where i can eliminate the 4th invites |
| 07:25.08 | grandjava | and stop asking asterisk to send invites after the calls have been bridged |
| 07:26.27 | *** join/#asterisk shuric (alexander@alexander.office.inter-telecom.net.ru) |
| 07:28.10 | puppet | *starting to build on his interface to redirect calls since there doesnt exist a good one |
| 07:29.55 | *** join/#asterisk sskyles (~sskyles@155.205.205.68.cfl.res.rr.com) |
| 07:29.57 | grandjava | anyone???? |
| 07:30.49 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 07:32.13 | *** part/#asterisk zignig (~simon@203.217.15.10) |
| 07:32.38 | sskyles | Is there such a thing as a command-line driven IAX client that I can compile in Linux? |
| 07:33.09 | Jas_Williams | grandjava, do you have canreinvite=no in your sip.conf if you are behind nat you need it |
| 07:33.25 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
| 07:33.26 | *** part/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
| 07:33.47 | *** part/#asterisk mrgoby (~mrgoby@141.211.162.97) |
| 07:34.00 | grandjava | yeah |
| 07:34.07 | grandjava | no |
| 07:34.15 | grandjava | my canreinvite=yes |
| 07:34.19 | grandjava | and nat=yes |
| 07:34.28 | grandjava | we dun wan the rtp ports goin thru the server |
| 07:34.38 | Jas_Williams | change canreinvite to no sip reload and test |
| 07:34.42 | grandjava | as it will choke the hell out of the server if the user grows |
| 07:35.01 | grandjava | if i put the canreinvite=no...it will definately work |
| 07:35.18 | grandjava | but we wouldn't wan the stress the bandwidth |
| 07:35.48 | grandjava | is there anyway that we can eliminate the 4th invites |
| 07:36.03 | Jas_Williams | ah so that is your problem then :) get a sip aware router that will do all of the ipaddress changes or place asterisk on an non natted ipaddress |
| 07:36.09 | grandjava | cause that is the invites which causes the whole darn problem |
| 07:36.23 | grandjava | our asterisk is on real IP |
| 07:36.28 | grandjava | and is not behind a NAT |
| 07:36.55 | grandjava | but i believe...if we can solve this...it would help a lot of ppl out there |
| 07:37.26 | Eight | Ok, y'know what... |
| 07:37.35 | Eight | I'm thinking maybe it's time for me to brush up on my C... |
| 07:37.49 | grandjava | yeah |
| 07:37.57 | Jas_Williams | OK Is the 4th invite is the reinvite, One of the clients is not sending it's external address it is not a problem with * |
| 07:38.01 | Eight | and add a config option to asterisk (on by default) that will prevent peering when one of the addresses is in 'localnet' and one of them isn't. |
| 07:38.13 | Eight | because that seems to be a huge percent of the issues people are having. I had it myself. |
| 07:38.24 | grandjava | i've added localnet |
| 07:38.35 | grandjava | 192.168.0.0/255.255.0.0 |
| 07:39.00 | Eight | grandjava: ya, but that doesn't do anything with the reinvite stuff. |
| 07:39.01 | grandjava | that means to server all ip 192.168.XXX.XXX |
| 07:39.05 | zapa | hi all its me again i have echo in one side of my * the side of my ip phone, but if i make a call trough pstn the pstn caller don´t hear any echo just the ip phone in my network any clue |
| 07:39.28 | *** join/#asterisk Alexis (~alexis@www.trim.it) |
| 07:39.32 | Alexis | hi |
| 07:39.46 | grandjava | the internal ip is send by a* |
| 07:40.04 | grandjava | basically...both calls can go thru... |
| 07:40.18 | grandjava | if the 4th invite is not there |
| 07:40.31 | grandjava | u wan to know y... |
| 07:40.32 | ikey | hi can any one guide me to configure asterisk as h323 gateway? |
| 07:41.38 | puppet | hmm |
| 07:41.51 | *** join/#asterisk JumpManJr (~guess@24-116-128-80.cpe.cableone.net) |
| 07:42.13 | grandjava | eight |
| 07:43.07 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
| 07:43.51 | puppet | hmm |
| 07:43.58 | puppet | Im having same problem asmrgoby |
| 07:45.33 | JumpManJr | I've never set up one of these before. However, I'm thinking of setting up with Broadvoice or Lingo. Any comments that might help. I've been scouring through the manuals and what not and I think I'm ready to give it a go. Just thought I would come here and see if anyone had any additional comments. |
| 07:45.36 | xeet2 | zapa: common. look up echo on voip-info.org |
| 07:46.56 | zapa | thanks xeet2 |
| 07:51.07 | SexyKen | Can't anyone help me out with my cdr issue :-\ |
| 07:52.06 | puppet | Redirect ROCKS |
| 07:52.27 | puppet | http://sip.puppet.nu/sip/test2.php?channel=SIP/1000-e386&exten=s&context=queuemusic&prio=1 works nice ;p |
| 07:52.31 | SexyKen | I wish my shit worked! gRRRRRRRRRRRRRRRR |
| 07:52.51 | puppet | now i know what prio does ;D |
| 07:52.57 | puppet | it is where it starts in the context |
| 07:53.00 | puppet | ^^ |
| 07:53.32 | puppet | or not ;p |
| 07:54.34 | *** join/#asterisk Cresl1n (~Cresl1n@251.134.218.209.transedge.com) |
| 07:57.57 | Inv_arp | JumpManJr: give it a go |
| 07:58.16 | Inv_arp | SexyKen: what probs ya have |
| 07:58.43 | nine76 | heh,so does a x100p work with astwin? See I convinced a friend to order me a x100p, he did,and ordered himself one as well. |
| 07:58.59 | Inv_arp | astwin? whats that |
| 07:59.01 | SexyKen | Inv_arp, I need a way to track calls in Asterisk...like for instance, cdr doesn't 'group' records together that asterisk created from the same original spawned call |
| 07:59.02 | drumkilla | nine76: no |
| 07:59.05 | nine76 | ouch |
| 07:59.08 | nine76 | hes screwed:) |
| 07:59.19 | nine76 | Seems I have 2 for my own play then |
| 07:59.42 | drumkilla | tell him to strap on a pair and install linux :) |
| 08:00.12 | nine76 | Ive been trying that for 2 years now. |
| 08:00.15 | nine76 | Hes skurd |
| 08:01.23 | Inv_arp | SexyKen: use a cdr_odbc or any other db |
| 08:01.44 | ikey | can any one help in configuring h323 gateway with asterisk |
| 08:02.09 | SexyKen | Inv_arp, I'm using mySQL. |
| 08:02.24 | SexyKen | Inv_arp, Calls still aren't grouped by where they started. |
| 08:02.42 | SexyKen | I need a way to link each 'spawned' call back to where it originally came from |
| 08:03.05 | puppet | iik when Im done with all single phpfiles i have to do parsing and make it to one interface ;p |
| 08:06.27 | *** join/#asterisk mikey-l (mikey@209.121.79.35) |
| 08:08.51 | *** join/#asterisk jmhunter (~jmhunter@64.77.199.223) |
| 08:08.51 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 08:09.08 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 08:09.30 | Eight | Question: How do I see this in the console? ast_log(LOG_DEBUG, "Deferring reinvite on '%s'\n", p->callid); |
| 08:10.07 | Eight | It happens at channel creation, so I can't do 'debug channel <channel>' because it doesn't exist yet, and once it does the messages has already been sent. |
| 08:12.27 | jmhunter | has anyone at von seen Damin, (greg) recently? |
| 08:14.02 | djin | ~seen Damin |
| 08:14.04 | jbot | damin is currently on #asterisk. Has said a total of 6 messages. Is idling for 1h 33m 45s |
| 08:15.21 | *** mode/#asterisk [+o brc_] by jmhunter |
| 08:15.57 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:18.07 | *** part/#asterisk DaLion (DaLion@Toronto-HSE-ppp3770824.sympatico.ca) |
| 08:19.23 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
| 08:24.41 | grandjava | what would be the sip call flows when both the clients are behind different NAT and asterisk is resolving the clients wan media ports but itself by requesting the clients to send their rtp to the asterisk |
| 08:27.01 | *** part/#asterisk buddah (~buddah@66-214-239-6.lb-cres.charterpipeline.net) |
| 08:34.37 | Eight | grandjava: uncomment the ;debug debug; line in logger.conf, restart asterisk. |
| 08:34.48 | Eight | then % tail -f /var/log/asterisk/debug; |
| 08:35.06 | Eight | note, your console only updates when the write buffer gets flushed, so it'll be a bit delayed. |
| 08:37.44 | grandjava | ok... |
| 08:37.47 | grandjava | got it... |
| 08:39.44 | SexyKen | Inv > Guess you dont have any ideas for help eh? |
| 08:42.32 | opus___ | hmmm |
| 08:42.42 | opus___ | does anyone have problems calling out on broadvoice, I think my config is wrong |
| 08:43.01 | opus___ | sip show channels shows TX ACK and RX: INVITE |
| 08:43.13 | opus___ | for the correct device and for 147.135.0.128 |
| 08:43.30 | Eight | opus___: my BV is working fine at the moment, using sip.broadvoice.com |
| 08:44.46 | opus___ | exten=_1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) |
| 08:44.46 | opus___ | exten=_1NXXNXXXXXX, 2, congestion() ; No answer, nothing |
| 08:44.48 | opus___ | exten=_1NXXNXXXXXX, 102, busy() ; Busy |
| 08:44.51 | opus___ | (sorry) |
| 08:45.09 | jmhunter | something so satisfying about watching a flaming moth fly away from a bug zapper |
| 08:45.10 | opus___ | SHouldn't be anything wrong with that, .. i hope. i have [sip.broadvoice.com] |
| 08:45.44 | Qwell | opus___: I love the comment on line 3 |
| 08:46.01 | opus___ | yay.. |
| 08:47.33 | Eight | opus___: I don't dial a 1 first on my calls through BV. |
| 08:47.47 | Eight | opus___: ${EXTEN:1} would change that. |
| 08:48.01 | Eight | opus___: I don't know if that's an issue or not, but that's how I have it set up. |
| 08:48.10 | puppet | ~seen puppet |
| 08:48.11 | jbot | puppet is currently on #asterisk (1d 11h 36m 26s) #debian (1d 11h 36m 26s). Has said a total of 316 messages. Is idling for 1s |
| 08:48.19 | Eight | ~seen eight |
| 08:48.20 | jbot | eight is currently on #asterisk (1d 18h 23m 28s). Has said a total of 321 messages. Is idling for 1s |
| 08:48.26 | puppet | bah |
| 08:48.28 | Eight | =) |
| 08:48.30 | Qwell | ~seen Qwell |
| 08:48.31 | jbot | qwell is currently on #asterisk. Has said a total of 161 messages. Is idling for 1s |
| 08:48.32 | puppet | eight: you have more msges then me :/ |
| 08:48.36 | Qwell | I don't talk much. |
| 08:48.37 | puppet | eight: you suck :( |
| 08:48.45 | puppet | eight: you dont have a life since you have more then me! ;D |
| 08:49.02 | Eight | puppet: That I'll concede =p |
| 08:49.05 | Qwell | Does that mean I have twice the life he does? :p |
| 08:49.07 | puppet | ;D |
| 08:49.11 | puppet | qwell: no |
| 08:49.12 | puppet | qwell: ;D |
| 08:49.35 | Eight | incidentally: C code sucks. |
| 08:49.37 | puppet | http://sip.puppet.nu/sip/test.php < phat |
| 08:49.41 | Eight | No wonder I haven't used it in years. |
| 08:49.50 | puppet | have parsed it some now ;p just gone remove the pings and then im gone do the heavy parsing |
| 08:51.03 | opus___ | 147.135.0.128 5034813752 506cabad1ef 00104/00000 ulaw (d) Tx: ACK |
| 08:51.06 | opus___ | hmmm |
| 08:51.09 | opus___ | thats as far as it goes |
| 08:51.39 | opus___ | well, i'll figure it out tomorrow, good night |
| 08:53.16 | Juggie | puppet, parseing * output is a hassle |
| 08:53.17 | *** part/#asterisk jmhunter (~jmhunter@64.77.199.223) |
| 08:53.17 | Juggie | i gave up |
| 08:53.33 | Juggie | i just wrap all the manager output in <pre> </pre> and let the browser handle it |
| 08:55.49 | puppet | juggie: i have noticed ;P |
| 08:56.01 | puppet | juggie: but i have to to be able to make a good interface to redirect calls |
| 08:57.09 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 08:57.13 | *** join/#asterisk Delmar (~Delmar@222-152-57-78.adsl.inspire.net.nz) |
| 08:57.24 | Delmar | echo echo echo echo echo. :( |
| 08:57.31 | puppet | juggie: you know how i show logged in users? |
| 08:57.35 | Delmar | thats my rant for today. |
| 08:57.41 | Delmar | hows everyone else? lol |
| 08:57.51 | Delmar | puppet sip users? |
| 08:57.55 | Delmar | sip show peers |
| 08:57.59 | Delmar | iax ... |
| 08:58.03 | Delmar | iax2 show peers |
| 08:58.53 | puppet | delmar: yes |
| 08:58.58 | puppet | not peers |
| 08:58.59 | Delmar | :P |
| 08:59.04 | puppet | oh wai |
| 08:59.04 | Delmar | users? |
| 08:59.07 | puppet | wait ;D |
| 08:59.08 | puppet | thanks ;D |
| 08:59.10 | Delmar | what kinda users lol |
| 08:59.14 | Delmar | no problem :P |
| 08:59.22 | puppet | this is kinda a hell :/ |
| 08:59.26 | puppet | I cant see if they are a friend |
| 08:59.29 | Delmar | dude.. users on your p0rn ftp site aint part of asterisk :P |
| 08:59.31 | Delmar | hehe |
| 08:59.35 | Delmar | oh ok |
| 08:59.39 | puppet | Maybe can take it on Dyn D |
| 08:59.50 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 08:59.50 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 08:59.54 | puppet | if its Dyn D then I can redirect to it if port != 0 |
| 09:00.04 | Delmar | :P |
| 09:00.06 | *** join/#asterisk mesi (~player@dsl-082-083-150-235.arcor-ip.net) |
| 09:00.08 | bkw_ | yet again... free wifi |
| 09:00.10 | puppet | cause if port != 0 then its up ;p |
| 09:00.14 | bkw_ | WASABI |
| 09:00.16 | Delmar | bkw wtf? |
| 09:00.17 | puppet | bkw_: hahaha |
| 09:00.24 | puppet | bkw_: free wifi rocksor |
| 09:00.36 | Delmar | if its legitimate free yes. |
| 09:00.44 | puppet | delmar: pfft ;p |
| 09:00.46 | bkw_ | riight |
| 09:00.49 | puppet | delmar: if its open its legal |
| 09:00.53 | puppet | ;p |
| 09:00.54 | Delmar | not here in NZ. |
| 09:00.57 | bkw_ | or if you can sql inject it |
| 09:01.01 | puppet | hahahhahaa |
| 09:01.07 | Delmar | the laws changes here.. if u access it just because its open.. you go to jail. |
| 09:01.17 | puppet | lol |
| 09:01.19 | bkw_ | Delmar,where? |
| 09:01.21 | Delmar | if you walk into someones house just because the door is unlocked.. what happens? |
| 09:01.21 | puppet | i think its like that here in sweden to |
| 09:01.23 | puppet | but who knows? |
| 09:01.24 | Delmar | you go to JAIL. |
| 09:01.27 | Delmar | New Zealand. |
| 09:01.31 | mesi | Delmar: Really to jail? |
| 09:01.52 | mesi | Delmar: Ask, what happens, if you DONT walk in there. |
| 09:01.55 | Delmar | go read the crimes amendmend act relating to IT crimes. |
| 09:02.17 | Delmar | do not go to jail and collect $200 |
| 09:02.20 | Delmar | :P |
| 09:02.48 | mesi | Delmar: Well, suppose there is somebody in there. He just came in from shopping and had a heart attack. He lost consciousness and you, just because the door is open you don't get in, won't see him and thus can't help him! |
| 09:03.15 | Delmar | seriously.. the law changed not long ago and its so broad that.. if someone on here asks me to do work on their computer... remotely... id have to fax them a form to release me of all liability and to confirm that others have access to the system and all kinds of crap... |
| 09:03.32 | Delmar | i would never do remote work on someones box without something in writting.. not now. its fucked. |
| 09:03.56 | mesi | Delmar: I would like this somebody to go in my house and help me, and if I just forgot to lock the door, I would like him to go in and look if there's somebody and if not do something so that no real theaf can come in and rob me! |
| 09:03.59 | puppet | Hmm |
| 09:04.28 | puppet | show channels always return first SIP/ or IAX/ or Zap/ |
| 09:04.31 | puppet | right? |
| 09:04.33 | mesi | Delmar: Well, I have to admit that all this just sounds like germany here :-) |
| 09:04.35 | Delmar | mesi, well if i walk along the street and see someones door open .. how am i meant to know someones has had an accident... im meant to look into some fuckin crystal ball and enter someones house just because their door is open? |
| 09:04.37 | Delmar | lol |
| 09:04.52 | Delmar | puppet u do... sip show channels |
| 09:04.54 | Delmar | or whatever |
| 09:05.19 | mesi | Delmar: Ok, ok. Anyway, your law sounds much too strict to me. But it is quite the same here in Germany. |
| 09:05.28 | Delmar | mesi, dude.. i could almost have someone in court on charges for doing a port scan on my box. |
| 09:05.32 | Delmar | its pretty bad here now. |
| 09:05.56 | Delmar | but since they didnt actually access anything.. thats a bit harder... |
| 09:06.17 | mesi | Delmar: Yes, that's brainwashed. |
| 09:06.36 | Delmar | the law is somewhat broad here is all. |
| 09:06.58 | Delmar | unsolicited email could almost be .. unlawful access of a computer system.. which is a charge. |
| 09:07.13 | mesi | Delmar: We have a law that enforces everybody running a website to put his first name, last name, telephone number, fax number, email address and additional information depending on his job online. |
| 09:08.09 | Delmar | mesi, hrm. you have to understand why they have done it tho. |
| 09:08.19 | mesi | Delmar: Nobody is interested in the fact, that data submission on the net is clear text for those websites and somebody could easily fake all this or erase it during transmission to the website user. |
| 09:08.21 | Delmar | sure they are going overboard... and ther must be a comprimise. |
| 09:08.45 | Delmar | not clear text if you know how to encode it right. |
| 09:08.54 | Juggie | delmar, "ohhh canada.... " |
| 09:08.56 | mesi | Delmar: Well, for commercial websites it might be ok, but for a personal homepage... |
| 09:09.22 | Delmar | still sounds a bit fucked to me tho |
| 09:09.25 | mesi | Delmar: I mean cleartext, it is not encrypted by strong cryptographic algorithms. |
| 09:11.08 | Delmar | right |
| 09:11.20 | Delmar | i think the owners details should be on any website |
| 09:11.31 | Delmar | but the designer...an individual..no way |
| 09:11.35 | Delmar | thats personal. fuck off. |
| 09:12.01 | modulus_ | Delmar, do you like chicken? |
| 09:12.27 | Delmar | prefer a good chunk of steak on the bbq myself. why |
| 09:12.40 | modulus_ | just noticed you're in a sharing mood |
| 09:12.48 | puppet | woho! |
| 09:13.00 | Delmar | lol |
| 09:13.05 | puppet | http://sip.puppet.nu/sip/test.php just got some left |
| 09:13.22 | puppet | now it puts IAX2/IAX/SIP/Zap into a new array ;p |
| 09:13.38 | puppet | now im gone devide info correct ;o) this is gone be so ugly code ;D |
| 09:13.43 | Juggie | puppet, your still going to have to write to parse EVERY command |
| 09:13.46 | Juggie | its ugly |
| 09:14.37 | puppet | juggie: i know... |
| 09:14.39 | puppet | juggie: :/ |
| 09:14.44 | puppet | juggie: but someone have to do it right? ;P |
| 09:15.00 | puppet | juggie: if people want ugly code ill release it |
| 09:15.10 | puppet | it does it job notmore |
| 09:16.37 | puppet | juggie: http://pastebin.ca/7230 |
| 09:17.28 | puppet | gnoP :esnopseR !!! |
| 09:17.28 | puppet | ;D |
| 09:18.22 | Juggie | puppet, i gave up and just wrapped it in <pre> </pre> |
| 09:18.29 | Juggie | until theres a better api its too hard to work with |
| 09:18.40 | Juggie | xml would be nice, but mark is totally against it |
| 09:19.14 | puppet | juggie just to be able to get connections without command would rock |
| 09:22.30 | *** join/#asterisk plovs_work (~plovs@195.13.248.82) |
| 09:24.57 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 09:26.12 | plovs_work | anybody know how to solve this: asterisk@home with x-lite on mac and windows, all phones give "nobody is available to take your call" |
| 09:30.37 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:33.46 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 09:41.44 | Delmar | bah. gotta be my phone line is out of balance. there is no other explanation for this lopsided echo bull shit. |
| 09:42.05 | Delmar | there is nothing more I can do. you name it.. i have done it. echo echo echo echo. |
| 09:43.12 | Delmar | fuck i could use two tin cups and some string and get better results. cant see myself transferring calls easily with that idea tho huh. LOL |
| 09:46.45 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 09:47.24 | *** join/#asterisk SirPrize (~blah@host-83-146-24-221.bulldogdsl.com) |
| 09:47.59 | SirPrize | How can I invoke something that would cause Asterisk to make TWO calls out, and then connect these two calls together ? |
| 09:49.36 | grandjava | drop a call file |
| 09:50.51 | puppet | I rock! |
| 09:50.52 | puppet | ;D |
| 09:50.54 | puppet | http://sip.puppet.nu/sip/test.php |
| 09:50.55 | puppet | pwned |
| 09:51.12 | Delvar | SirPrize: look at /usr/src/asterisk/sample.call it says prety much what it does int eh file |
| 09:51.18 | SirPrize | I know I can make a call via a call, file, but how do I connect/bridge the two calls that are made via these call files ? |
| 09:51.53 | Delvar | SirPrize: its automatic |
| 09:52.32 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:52.39 | Delvar | SirPrize: send Channel: to first line, then set Application: to Dial and set the variables to SIP/bla/bla |
| 09:52.54 | Delvar | puppet: ok so what is that? |
| 09:53.24 | SirPrize | Delvar .... ok .... , but maybe I'm misunderstanding how to bridge the two calls then |
| 09:53.58 | SirPrize | Delvar: Ah, ok. I got it now |
| 09:54.34 | Delmar | hrm. |
| 09:54.34 | puppet | this is just retarded to try and parse it :( |
| 09:54.37 | Delvar | SirPrize: cool, i use call files all the time, mainly for testing, they can be VERY usfull |
| 09:55.41 | Delmar | assuming that its not the zaptel/asterisk software behind these echo problems.. I mean.. I have tried everything one could imagine... and assuming for jsut one second that its not the X100p, then its back to the phone line. tomorrow im gonna strip the line back so that there is nothing but 2 wires into the card from the exchange. |
| 09:55.49 | Delvar | puppet: what are you trying to do, looks kinda interesting |
| 09:55.56 | Delmar | maybe then I will kick myself and figure out .. what.. on my line.. is the cause.... |
| 09:56.04 | Delmar | OR .. i will prove the card is at fault. |
| 09:56.06 | SirPrize | Delvar: Yeah - it's pretty neat! Am just getting into it, and find it totally fascinating what can be accomplished with it |
| 09:56.13 | puppet | delvar: http://pastebin.ca/7231 |
| 09:56.18 | puppet | delvar: thats what im trying to do |
| 09:56.20 | puppet | parse it :/ |
| 09:56.37 | puppet | but its just retarded think i have to do some strange regexp with all kinda alternativies that can happen |
| 09:58.23 | Delvar | puppet:i see, looks like fun :) |
| 09:58.30 | puppet | delvar: NO ;p |
| 09:58.38 | puppet | but When I get it to work then its gone be fun fun fun |
| 09:58.49 | puppet | then I can have a php page with active channels and connect as i want |
| 09:58.52 | puppet | php based switchboad |
| 09:59.18 | Delvar | puppet: think about modifying the chow channels command to output easyer to parse text? |
| 09:59.31 | puppet | delvar modify how where? |
| 09:59.41 | Delvar | puppet: the source code |
| 09:59.41 | puppet | would be super to have it in CSV;P |
| 09:59.44 | puppet | iik |
| 09:59.47 | Delvar | exactly |
| 09:59.50 | puppet | bah dont wanne touch the sourcecode |
| 09:59.54 | puppet | :/ |
| 10:00.07 | puppet | is it hard to make a module? |
| 10:00.15 | Delvar | show channels csv (would be cool) |
| 10:00.25 | Delvar | if you know C then it isnt too hard |
| 10:00.34 | puppet | im gone look in the code |
| 10:00.36 | puppet | i dont know C tho |
| 10:00.38 | puppet | brb |
| 10:00.53 | Delvar | haha |
| 10:01.28 | SirPrize | Is there any way for callfile triggered calls to know when the remote channel end answered, and only THEN to start executing the rest of the dialplan? |
| 10:03.40 | *** join/#asterisk xyharley (~daecon@xyharley.dsl.xmission.com) |
| 10:05.20 | puppet | lol |
| 10:05.27 | puppet | not even finding the code for the printout |
| 10:05.27 | *** part/#asterisk SirPrize (~blah@host-83-146-24-221.bulldogdsl.com) |
| 10:05.47 | Eight | Ugh... C.... |
| 10:05.55 | Eight | I hate this language. |
| 10:06.01 | Juggie | something totall offtopic whats the best drive image software |
| 10:06.19 | puppet | ANyone that know the * source here? |
| 10:06.25 | Eight | puppet: I wish =/ |
| 10:06.39 | puppet | I just wanne add show channels CSV |
| 10:06.41 | puppet | :/ |
| 10:06.52 | puppet | would make life so much easier :( |
| 10:07.26 | Eight | I don't know if it's the asterisk source, or just lack of experience with C, but this is really a pain to deal with. |
| 10:07.44 | Eight | I haven't worked on anything beyond trivial without good OO in a long time. |
| 10:09.07 | puppet | found it |
| 10:09.20 | puppet | IT EXISTS |
| 10:09.25 | puppet | show channels concise |
| 10:09.30 | puppet | !!!! ;D;D;D;D |
| 10:09.58 | Eight | heh, just not in the help? |
| 10:10.27 | *** join/#asterisk joaovianna (joao@node-40247a6a.ewr.onnet.us.uu.net) |
| 10:10.37 | puppet | Nope |
| 10:10.40 | puppet | read it in the code |
| 10:10.44 | puppet | : delimited |
| 10:11.21 | puppet | ast_cli(fd, CONCISE_FORMAT_STRING, c->name, c->context, c->exten, c->priority, ast_state2str(c->_state),c->appl ? c->appl : "(None)", c->data ? ( !ast_strlen_zero(c->data) ? c->data : "" ): "",(c->callerid && !ast_strlen_zero(c->callerid)) ? c->callerid : "",(c->accountcode && !ast_strlen_zero(c->accountcode)) ? c->accountcode : "",c->amaflags); |
| 10:11.37 | Juggie | puppet, doesnt exist for all does it? |
| 10:11.43 | puppet | yes it does |
| 10:11.46 | puppet | :) |
| 10:12.03 | Juggie | not working for me |
| 10:12.28 | Juggie | maybe it has to be called from the manager api |
| 10:12.48 | puppet | no |
| 10:12.52 | puppet | wich version you got? |
| 10:13.00 | Juggie | both |
| 10:13.02 | Juggie | head & stable |
| 10:13.29 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 10:13.36 | puppet | 1.0.6 |
| 10:13.46 | puppet | show channels concise |
| 10:14.01 | Juggie | returns nothing for me |
| 10:14.05 | Juggie | of course i have no channels up |
| 10:14.12 | puppet | thats why |
| 10:14.26 | puppet | its empty set if no channels are up |
| 10:14.28 | Juggie | but for example sip show peers |
| 10:14.28 | puppet | like it should be |
| 10:14.32 | Juggie | sip show peers concise |
| 10:14.33 | Juggie | does nothing |
| 10:16.10 | puppet | http://sip.puppet.nu/sip/test.php |
| 10:16.16 | Juggie | yah i see it |
| 10:16.19 | puppet | does just work on channels |
| 10:16.20 | Juggie | for show channels it works |
| 10:16.20 | puppet | :/ |
| 10:16.28 | Juggie | that sucks |
| 10:16.29 | puppet | i want one for peers to |
| 10:16.32 | Juggie | if it worked on everything |
| 10:16.33 | puppet | i want concine toALL |
| 10:16.35 | Juggie | that would be ideal |
| 10:18.07 | puppet | http://sip.puppet.nu/sip/test.php |
| 10:18.07 | puppet | ;D |
| 10:18.09 | Eight | wtf is ast_channel defined? |
| 10:18.25 | Eight | I'm just no good at grepping .c file s=p |
| 10:18.28 | puppet | IT WORKS ;D |
| 10:18.30 | puppet | <3 |
| 10:18.33 | puppet | eight: cli.c |
| 10:19.23 | Eight | you lie. |
| 10:19.50 | puppet | nope |
| 10:19.51 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 10:20.08 | Eight | why would it be in cli.c anyways? |
| 10:20.25 | puppet | i found it there ;p |
| 10:20.46 | Eight | struct ast_channel { .. } ? |
| 10:20.58 | puppet | nah just found it in use ;p |
| 10:21.08 | Eight | ya, I wanted to know where the struct was define d=p |
| 10:21.38 | puppet | ;p fu ;D |
| 10:22.49 | *** join/#asterisk fotang (~mike@62.90.253.51) |
| 10:23.34 | puppet | http://pastebin.ca/7233 |
| 10:23.47 | Juggie | puppet, write up a list of commands which should have concise |
| 10:23.48 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 10:24.12 | Juggie | pastebin them and msg me the link |
| 10:24.27 | puppet | juggie ok |
| 10:24.37 | Juggie | my c is good enough to write those mods |
| 10:24.49 | puppet | juggie: its really all show tho :/ |
| 10:25.06 | puppet | since its show you use to get data |
| 10:25.09 | Juggie | yes but only a few exist in cli.c |
| 10:25.19 | Juggie | show channels, show modules etc... |
| 10:25.26 | puppet | Ill see :) |
| 10:25.32 | Juggie | most of them are in modules like chan_sip etc |
| 10:25.33 | Juggie | like |
| 10:25.35 | Juggie | sip show peers |
| 10:25.56 | Delvar | puppet: cool! i didnt even know about it |
| 10:26.10 | Delvar | puppet: your script working better now? |
| 10:26.53 | puppet | delvar: working perfect |
| 10:27.08 | puppet | on next step now :) |
| 10:27.13 | puppet | but before that im gone get juggie a list |
| 10:27.13 | Delvar | pippet: see looking a the source was a good idea :) |
| 10:27.27 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 10:27.31 | Mother_ | hi all |
| 10:27.59 | puppet | juggie: show channel maybe should haev a csv output? |
| 10:28.02 | Delvar | you neever know what they have put in it |
| 10:28.45 | *** join/#asterisk vladuk (~vk@mustang.bzcrew.com) |
| 10:29.02 | puppet | delvar: fun thing ;P i have really highlighter pippet to ;p |
| 10:29.10 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 10:29.17 | Juggie | puppet, : is fine we'll stick with that |
| 10:29.28 | Delvar | hehe oops :) |
| 10:29.35 | vladuk | hi ALL |
| 10:29.54 | puppet | delvar: http://sip.puppet.nu/sip/test.php there is the result now |
| 10:29.55 | vladuk | had anyone experience with asterisk cross compiling ? |
| 10:30.10 | puppet | delvar: want the source to it? |
| 10:30.11 | Juggie | write up all the commands that should be csv, what i will do is start on a few patches, and when i get enough done, i will post it to the bug tracker, and ask for people to take a module, and apply concise to it. |
| 10:30.33 | Delvar | puppet: i wouldnt say no to free code :) |
| 10:30.46 | puppet | juggie: in cli.c its just show channel ill check sip.c now |
| 10:31.22 | Juggie | its alot of code to mod, so if a bit was done, then post to tracker you might get people willing to work on it |
| 10:31.24 | Juggie | since its easy stuff |
| 10:31.30 | puppet | http://pastebin.ca/7234 |
| 10:31.40 | Juggie | priv message it to me |
| 10:31.47 | Eight | Hrmm, extension 'i' isn't working as I expect it. |
| 10:32.06 | Juggie | i is invalid, whats the issue |
| 10:32.16 | Eight | if I dial in with x-lite to an extension that doesn't exist, I get 404, not invalid. |
| 10:32.19 | Delvar | cool thanks |
| 10:32.42 | Eight | not the extension 404, but the SIP code 'not found'. |
| 10:33.28 | fotang | i got a cisco ata186 (IP: 192.168.0.205) with a user connected and registered with asterisk (192.168.0.253). however, `sip show peers` displays the host as being the asterisk server (192.168.0.253), and not the ata186. softphones connecting from PC's show up with the correct hosts. this is my first attempt at settin up asterisk, so was just wondering... |
| 10:33.48 | fotang | that was quite a mouthful! |
| 10:34.12 | Delvar | puppet: ill find a use for that later :) |
| 10:35.23 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 10:35.56 | puppet | delvar: im sure you will ;p |
| 10:37.22 | Eight | Ok, I know this can't possibly be that difficult. |
| 10:37.31 | Eight | Is there a primer for how the asterisk code is laid out, anywhere? |
| 10:38.22 | Eight | Y'know... I wonder if it would be too hard to rewrite a bunch of the higher level logic in python... |
| 10:38.34 | Eight | and leave the C code to just do the heavy lifting. |
| 10:38.53 | puppet | eight: why you want to do that? ;P |
| 10:39.01 | puppet | python is python |
| 10:39.24 | Eight | puppet: because python is a far better language than C for things other than 'heavy lifting' =p |
| 10:39.33 | Eight | puppet: and it calls C cleanly. |
| 10:39.34 | puppet | pfft |
| 10:39.38 | puppet | php > python |
| 10:39.39 | puppet | ;D |
| 10:39.44 | Eight | PHP is *great* for web pages. |
| 10:39.50 | Delmar | hey.. anyone else here using a 2.6.x kernel and loading wcfxo modules? |
| 10:39.50 | Eight | and moronic for just about anything else =) |
| 10:39.53 | puppet | PHP is great for all |
| 10:39.53 | puppet | ;D |
| 10:39.54 | Delmar | module*. |
| 10:40.21 | Eight | puppet: I knew someone who was actually reconstructing python pickled (archived) objects in PHP. |
| 10:40.35 | Eight | puppet: instead of writing 4 lines of python and calling that externally. |
| 10:40.47 | puppet | ;p |
| 10:40.54 | Eight | puppet: it seems PHP people really like to use it for everything =p |
| 10:40.59 | puppet | yeah yeah ;D <3 PHP cause I can it like nothing else ;P |
| 10:40.59 | ta[i]nted | is BV still have issues? |
| 10:41.17 | Eight | ta[i]nted: in what regard? It's working for me at the moment (if you exclude the chicago proxy). |
| 10:41.31 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 10:41.50 | ta[i]nted | i'm get some "Got SIP response 404 "not found" back from .." errors during calls |
| 10:43.01 | Eight | ta[i]nted: haven't had that, I don't think. Just 403 forbidden. |
| 10:43.12 | Eight | ta[i]nted: what's your dial command? |
| 10:43.31 | Eight | just verified BV is working for me, FYI. |
| 10:44.40 | Delmar | anyone here using an X100p card, and kernel 2.6.x? |
| 10:45.43 | Delmar | holey crap ilbc is a pig on resources. |
| 10:45.48 | puppet | I should learn how to write a class now ;p |
| 10:46.03 | *** join/#asterisk dubphil (~dubphil@80.124.137.201) |
| 10:46.16 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 10:46.36 | Eight | class? |
| 10:46.39 | puppet | ask coding class for managemer |
| 10:46.41 | puppet | afk* |
| 10:46.44 | puppet | class for php |
| 10:46.53 | Eight | ah. |
| 10:47.00 | Eight | I was going to say, ast' isn't OO =p |
| 10:47.25 | *** join/#asterisk Ryuujin (~ryuujin@core1.eqixva-gige.oceanius.com) |
| 10:50.48 | dubphil | Hello, I was searching all around to see how much phone lines we can manage (at the hardware level) with asterisk/digium ? |
| 10:51.05 | dubphil | I need about 200 lines |
| 10:51.40 | Eight | You want to run 8 T1s on a single box? |
| 10:51.56 | Ryuujin | Brumle :) |
| 10:52.05 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 10:52.29 | *** part/#asterisk christo (~chris@office.enovi.com) |
| 10:53.34 | Eight | well, I 'fixed' the broken invalid thing. |
| 10:53.40 | Eight | ; Leave this at the BOTTOM of the context to grab anything else |
| 10:53.40 | Eight | exten => _.,1,goto(i,1) |
| 10:53.42 | Eight | =p |
| 10:53.59 | dubphil | Eight: I missed something, what hardware are needed to connect 200 phones on an Asterisk system ? |
| 10:54.28 | Eight | dubphil: Analog phones? You'd need 2 Quad T1 cards and 8 channel banks. |
| 10:54.44 | Eight | that's 192 phones. |
| 10:54.51 | ta[i]nted | fucking broadvoice |
| 10:55.00 | Eight | ta[i]nted: what's your dial command? |
| 10:55.06 | ta[i]nted | my dial command/ |
| 10:55.08 | ta[i]nted | let me check |
| 10:55.27 | dubphil | Eight sorry I meant numeric phones |
| 10:55.38 | Eight | ... numeric phones? SIP Phones? |
| 10:55.41 | ta[i]nted | 1,Dial(SIP/broadvoice/xxxxxxxxxx,,t) |
| 10:56.01 | dubphil | What the difference ? |
| 10:56.11 | Eight | 1,Dial(SIP/##########@sip.broadvoice.com,t) |
| 10:56.47 | Eight | dubphil: I think they're the same thing, but I don't know what you think a 'numeric phone' is... |
| 10:57.05 | ta[i]nted | Eight does that make a difference? |
| 10:57.10 | Eight | ta[i]nted: I have no idea? |
| 10:57.40 | dubphil | <PROTECTED> |
| 10:57.41 | Eight | ta[i]nted: but you're the one getting the 404, and I'm not, so I thought I'd mention it =) |
| 10:58.13 | Eight | dubphil: well, if it's a bunch of SIP phones the main thing is going to be what codecs they're all using. |
| 10:58.23 | Eight | dubphil: If you're doing alot of transcoding your hardware requirements are going to be higher. |
| 10:58.32 | Eight | dubphil: RAM isn't much of a concern, CPU is. |
| 10:59.27 | dubphil | we use Alcatel phones 4010 4020 4035 |
| 10:59.44 | *** join/#asterisk doppiaemme (~max@131.175.126.186) |
| 10:59.55 | dubphil | is it compliant ? |
| 11:00.04 | Eight | dubphil: dunno. |
| 11:00.21 | puppet | http://sip.puppet.nu/sip/test.php |
| 11:00.23 | puppet | ;D |
| 11:00.26 | puppet | rockzor |
| 11:00.34 | Eight | slowzor =p |
| 11:00.41 | *** join/#asterisk shanky (~shanky@62.37.166.37) |
| 11:00.44 | shanky | hi |
| 11:00.56 | puppet | eight: since i suck onsockets i do so i reconnect at each request |
| 11:01.15 | Eight | Hehe, I love jedi-extension-trick.gsm |
| 11:01.15 | puppet | eight: and since the mangment service suck since its really single user ;O) its best ;D |
| 11:01.21 | shanky | is there any variable I could change in the sip.conf to put the volume of the calls up? |
| 11:01.46 | dubphil | what are channel banks ? where can I see them ? |
| 11:02.12 | Eight | dubphil: oh, I just realized what you meant by numeric vs analog vs SIP. |
| 11:02.41 | Eight | dubphil: are these 'internet' phones, or just normal phones that you can plug into an RJ11 jack? |
| 11:03.37 | dubphil | Eight: we can either plug on RJ11 or RJ45 depends on the cabling structure of the office |
| 11:03.56 | *** join/#asterisk jeffik (~jeffik@user-v7kajot.dialup.mindspring.com) |
| 11:04.50 | puppet | hahaha |
| 11:05.09 | puppet | delvar: do an help show channels |
| 11:05.10 | puppet | lol |
| 11:06.15 | [ro]nic3try | sal |
| 11:07.19 | [ro]nic3try | new day.. new problems :) |
| 11:07.35 | dubphil | I must leave, thanks Eight, I will come later if it doesn't disturb the channel :) |
| 11:08.37 | Ryuujin | I wants a 1mbit sdsl :( |
| 11:09.59 | *** part/#asterisk dubphil (~dubphil@80.124.137.201) |
| 11:11.06 | [ro]nic3try | is possible to have the rtp not going through astrisk ? |
| 11:12.04 | grandjava | yeah |
| 11:12.15 | grandjava | just change the canreinvite to yes |
| 11:18.27 | [ro]nic3try | not working :( |
| 11:21.25 | *** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com) |
| 11:22.58 | grandjava | wat is the problem |
| 11:23.04 | grandjava | when u say not working |
| 11:23.18 | Delvar | puppet: well, who looks at help files anyway! :) |
| 11:23.30 | [ro]nic3try | i still have the rtp send through asterisk |
| 11:23.41 | grandjava | how do u know |
| 11:23.47 | [ro]nic3try | i use ethereal |
| 11:23.52 | grandjava | ok... |
| 11:23.58 | grandjava | where do u put ur canreinvite?? |
| 11:24.20 | [ro]nic3try | in sip.conf at each phone |
| 11:24.27 | grandjava | that is right |
| 11:24.46 | grandjava | wierd it should work... |
| 11:25.08 | [ro]nic3try | but.. i woder.. culd this be happening because the phone is behing NAT ? |
| 11:25.18 | grandjava | do u put nat=yes |
| 11:25.33 | [ro]nic3try | yes |
| 11:25.47 | Mother_ | grrrr |
| 11:26.09 | grandjava | how is ur sip.conf setting looks like |
| 11:26.21 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 11:26.28 | Mother_ | I have recorded a gsm file on one asterisk box, sent it to another box, and it complains that |
| 11:26.29 | Mother_ | Mar 11 12:24:51 WARNING[2388]: file.c:486 ast_openstream_full: File abc.gsm does not exist in any format |
| 11:26.29 | Mother_ | Mar 11 12:24:51 WARNING[2388]: file.c:790 ast_streamfile: Unable to open abc.gsm (format unknown): No such file or directory |
| 11:26.38 | [ro]nic3try | it's allmoust like the default :) |
| 11:26.46 | Mother_ | when I have recorded and played the file no problem on the other box |
| 11:27.39 | [ro]nic3try | now.. i test that by call from a phone registered ro * to a phone on another SER server |
| 11:27.44 | Darwin35 | usr/bin/ld: cannot find -lodbc |
| 11:27.44 | Darwin35 | gmake[1]: *** [res_odbc.so] Error 1 |
| 11:28.30 | antifuchs | let me repeat that asterisk realtime rocks. |
| 11:28.51 | Darwin35 | /usr/bin/ld: cannot find -l/usr/local/lib |
| 11:28.51 | Darwin35 | gmake[1]: *** [res_odbc.so] Error 1 |
| 11:29.06 | Darwin35 | grrrr |
| 11:29.13 | antifuchs | I find that like realtime as much as I hated the old mysql thing in 1.0.1's chan_sip (: |
| 11:30.18 | grandjava | ok...where does the rtp cosmes from |
| 11:30.32 | grandjava | ser server or * server |
| 11:31.03 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 11:32.18 | [ro]nic3try | ser - > * then * -> ser |
| 11:32.37 | puppet | woot |
| 11:32.38 | puppet | http://sip.puppet.nu/sip/test4.php |
| 11:32.43 | [ro]nic3try | ser - > * then * -> phone |
| 11:32.46 | puppet | info about a channel ;p |
| 11:32.51 | *** part/#asterisk RoyK (~roy@80.239.107.80) |
| 11:33.13 | puppet | i so rock ;D |
| 11:33.38 | Mother_ | argh |
| 11:33.41 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 11:33.46 | RoyK | ~seen coppice |
| 11:33.47 | jbot | coppice <~chatzilla@134.195.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 2d 20h 51m 49s ago, saying: 'mafoo: yes'. |
| 11:33.47 | Darwin35 | fixed |
| 11:33.49 | Mother_ | I had the file extension in the playback command |
| 11:33.53 | Mother_ | works now |
| 11:37.02 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
| 11:37.08 | ta[i]nted | is broadvoice restricting simultaneous outgoing channels now? |
| 11:37.12 | Bonbon | guys, can anyone recommend a good rating engine |
| 11:37.44 | Bonbon | commercial or non-commercial |
| 11:39.14 | [ro]nic3try | grandjava: after the ACK, SER send RTP to * , then * -> ATA186 |
| 11:42.00 | Darwin35 | chan_zap.c: In function `pri_dchannel': |
| 11:42.00 | Darwin35 | chan_zap.c:9063: error: structure has no member named `forwardnum' |
| 11:42.36 | [ro]nic3try | if i use 2 phone , i hav |
| 11:42.38 | puppet | eight: still here? |
| 11:42.41 | clive- | /who *.za |
| 11:42.49 | [ro]nic3try | have the same problem |
| 11:46.11 | Darwin35 | ok this is a pain |
| 11:49.28 | Eight | puppet: just walked back in. |
| 11:50.06 | puppet | eight: wanne see some cool? |
| 11:50.53 | Darwin35 | chan_zap.c:9063: error: structure has no member named `forwardnum' |
| 11:51.00 | Darwin35 | <PROTECTED> |
| 11:54.44 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 11:55.35 | Eight | So, is it possible to tell BV what CallerID info to use when I'm making an outgoing PSTN call? |
| 11:57.41 | ta[i]nted | Eight` no |
| 11:57.58 | Eight | ta[i]nted: darn =/ |
| 11:58.22 | Eight | That'd be handy. |
| 11:58.48 | ManxPower | ~docs |
| 11:58.49 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 11:58.49 | ta[i]nted | "white house" <5555555555>? |
| 11:58.58 | ManxPower | Don't use quotes in callerid! |
| 11:59.11 | Eight | ManxPower: ? |
| 11:59.12 | ta[i]nted | |white house| <5555555555>? |
| 11:59.25 | ta[i]nted | white house <5555555555> |
| 11:59.37 | ManxPower | Eight, Some IP phones refuse to work if they get a call with quotes in the calleridname. |
| 11:59.51 | ManxPower | ta[i]nted white house <5555555555> <-- this is correct. |
| 12:00.26 | shanky | I'm using asterisk just with sip phones, the communication is right |
| 12:00.40 | shanky | but I have a lot of echo |
| 12:00.55 | ta[i]nted | shanky speak slower |
| 12:00.57 | shanky | what can I configure? |
| 12:00.58 | ManxPower | shanky, Then you have a problem. Echo is one of the hardest things to solve. |
| 12:01.04 | ta[i]nted | drag your vowels out |
| 12:01.04 | Mikrozaur | If i calling from iax2 client 4444 to 4444 astersk servis is killed. It's bug ? |
| 12:01.14 | ta[i]nted | liiiiiiiiiiiiiiiikeeeeeeeee thiiiiiiiiiiiiiiiis |
| 12:01.24 | ta[i]nted | kills the echo everytime |
| 12:01.26 | ManxPower | Mikrozaur, Anything that causes Asterisk to crash is a bug. |
| 12:01.52 | shanky | what codec do you recommend me? |
| 12:01.57 | Mikrozaur | What i'm can doinf for fix it ? |
| 12:03.05 | Darwin35 | g729 for best performance otherwise g711 and speex |
| 12:03.13 | Mikrozaur | http://pastebin.ca/raw/7236 |
| 12:03.14 | Eight | alright, I'm definitely using 1.0.6 with BV, and no special patching. |
| 12:03.22 | Eight | updated the wiki http://www.voip-info.org/tiki-index.php?page=Asterisk+settings+Broadvoice |
| 12:03.25 | shanky | Darwin35: ok, I'm using g729 |
| 12:03.34 | *** part/#asterisk datareactor (~datareact@203.81.192.53) |
| 12:03.39 | Darwin35 | do yu have a licesne for g729 |
| 12:03.52 | ManxPower | shanky, Look in the Wiki |
| 12:04.03 | Darwin35 | from digium and did you download the codec |
| 12:04.03 | shanky | Darwin35: yes the eMTA I'm using has |
| 12:04.15 | Darwin35 | ok |
| 12:05.06 | ManxPower | shanky, eMTA? |
| 12:07.29 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 12:08.38 | Eight | Grumble... http://lists.digium.com/pipermail/asterisk-users/2005-January/082611.html |
| 12:08.58 | *** join/#asterisk jeffik (~jeffik@user-v7kajot.dialup.mindspring.com) |
| 12:12.20 | puppet | Bah im not getting park to work |
| 12:12.46 | *** join/#asterisk l-fy (~diana@diana.null.ro) |
| 12:12.49 | l-fy | ~h323 |
| 12:12.50 | jbot | hmm... h323 is An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is heavily dependent on other ... |
| 12:13.26 | *** part/#asterisk l-fy (~diana@diana.null.ro) |
| 12:14.46 | Darwin35 | ~sex |
| 12:14.47 | jbot | I'm sexless |
| 12:14.55 | ta[i]nted | what's the best way to upgrade |
| 12:15.19 | Darwin35 | grab the latest .tar.gz ver and make it and install it |
| 12:15.29 | ta[i]nted | just rm /etc/asterisk, grab cvs head, make it and install it? |
| 12:15.31 | Darwin35 | but update all your installed packages first |
| 12:15.40 | Eight | wget <tarball> && tar -zxvf <tarball> && cd asterisk-1.0.6 && make && sudo make install |
| 12:15.48 | Darwin35 | never rm /etc/asterisk |
| 12:15.56 | Darwin35 | that will screw up your system |
| 12:16.03 | ta[i]nted | so just let it overwrite whatever is in there? |
| 12:16.09 | Eight | ta[i]nted: seems to work alright. |
| 12:16.12 | Darwin35 | it wont over write |
| 12:16.19 | Darwin35 | thats your configs you need |
| 12:16.30 | Eight | Darwin35: he's talking about the /usr/bin and whatnot, NOT etc |
| 12:16.34 | ta[i]nted | what about the binaries |
| 12:16.39 | ta[i]nted | yea |
| 12:16.43 | Eight | make install; will overwrite everything BUT /etc, afaik. |
| 12:16.51 | ta[i]nted | ok |
| 12:16.57 | Darwin35 | no |
| 12:16.59 | ta[i]nted | do u guys run cvs-head or 1.0.6 |
| 12:17.00 | Darwin35 | it will now |
| 12:17.08 | Eight | Darwin35: what will what? |
| 12:17.17 | Eight | ta[i]nted: 1.0.6, I despise running CVS versions =p |
| 12:17.51 | Eight | a'course... Asterisk doesn't actually have a unified Changelog, which makes my life unpleasant =p |
| 12:18.16 | Darwin35 | not make install but the 2nd command that comes up does |
| 12:18.31 | Darwin35 | make menus |
| 12:18.46 | Darwin35 | I think sdoes |
| 12:18.46 | ta[i]nted | hmm |
| 12:18.50 | ta[i]nted | never heard of make menus |
| 12:18.50 | Darwin35 | make examples |
| 12:18.55 | ta[i]nted | is that sample configs? |
| 12:18.58 | Darwin35 | yes |
| 12:19.15 | Darwin35 | that will overwrite /etc/asterisk |
| 12:19.15 | puppet | this here works wonders now |
| 12:19.39 | Eight | So don't make examples. |
| 12:19.44 | Eight | just make && sudo make install |
| 12:19.45 | *** join/#asterisk citats (~james@duff.gnuinter.net) [NETSPLIT VICTIM] |
| 12:19.45 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM] |
| 12:19.45 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) [NETSPLIT VICTIM] |
| 12:19.45 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) [NETSPLIT VICTIM] |
| 12:19.59 | Darwin35 | make make install and you should be fine |
| 12:20.07 | Darwin35 | with the 1.0.6 ver |
| 12:20.52 | Eight | Asterisk apparently evaluates the extensions file in order. |
| 12:21.15 | Eight | exten => _.,1,goto(i); works. |
| 12:21.27 | ta[i]nted | hopefully that fixes my BV issues |
| 12:21.27 | Eight | when I put it at the bottom, that is. |
| 12:21.43 | ta[i]nted | their tech support is on crack |
| 12:21.47 | Eight | ta[i]nted: what version were you running before? |
| 12:21.55 | ta[i]nted | let me check |
| 12:21.56 | Eight | ta[i]nted: you actually got ahold of their tech support? |
| 12:22.01 | Darwin35 | I like BV but still having issues with not reaching a live person |
| 12:22.01 | Eight | I've called a half dozen times, on hold up to 30 minutes. |
| 12:22.07 | Eight | and no replies from support@. |
| 12:22.09 | ta[i]nted | yea after 25 min wait |
| 12:22.10 | Darwin35 | yep |
| 12:22.25 | Eight | and their automated stuff takes forever. |
| 12:22.36 | ta[i]nted | Asterisk CVS-HEAD-02/12/05-15:01:06 |
| 12:22.37 | Eight | "you should receive a confirmation e-mail shortly" took over 12 hours. |
| 12:22.45 | Darwin35 | I just get a busy tone 80% of the time |
| 12:22.46 | ta[i]nted | heh |
| 12:22.48 | ManxPower | You should never use _. as an extensions.conf pattern. |
| 12:23.04 | Darwin35 | I dial 611 and get them but never get trew |
| 12:23.07 | ta[i]nted | i like how ManxPower pipes in like that |
| 12:23.15 | ta[i]nted | very reassuring that he's around |
| 12:23.29 | Eight | ManxPower: A) Why? B) Do you have an alternative way to 'catch' non-extensions? invalid doesn't work in this situation. |
| 12:23.49 | Darwin35 | ManX is our savior |
| 12:23.50 | ta[i]nted | are the zaptel and libpri packages even required? |
| 12:24.08 | ta[i]nted | i don't have any zaptel channels.. |
| 12:24.12 | Eight | ta[i]nted: well, zaptel helps with meetme (ztdummy), and I have no idea about libpri. |
| 12:24.16 | ManxPower | Eight, Asterisk calles various non-number extensions, like exten => h when a call hangs up. Do you really want the _. to be called when a call hangs up. Use _X. |
| 12:24.17 | Darwin35 | he is here to guide us when we are lost . Pick us up when we are down |
| 12:24.36 | Darwin35 | he guides us to the * promiss land |
| 12:24.39 | ta[i]nted | ManxPower are zaptel and libpri required? |
| 12:24.50 | ManxPower | Darwin35, like other "savior" nobody wants to give me a job 8-) |
| 12:24.52 | Darwin35 | not if your jsu doing sip |
| 12:24.52 | ta[i]nted | ManxPower is the * orb |
| 12:24.55 | ManxPower | ta[i]nted, For what? |
| 12:25.06 | Eight | ManxPower: aha, good deal. Thanks |
| 12:25.08 | ta[i]nted | ManxPower for an asterisk install |
| 12:25.20 | Darwin35 | if you are using a tdm or other card and hard lines tes |
| 12:25.21 | ManxPower | ta[i]nted, only if you want to use zaptel or PRI. |
| 12:25.30 | ta[i]nted | yea thought so |
| 12:25.46 | ta[i]nted | i remember reading a faq someplace where it said zaptel was required |
| 12:25.54 | Darwin35 | no |
| 12:25.56 | ta[i]nted | but they are wrong |
| 12:25.58 | ta[i]nted | wrong |
| 12:26.04 | ManxPower | ta[i]nted, It was wrong, or gave information specific to something else, like MeetMe. |
| 12:26.16 | ManxPower | MeetMe requires Zaptel. |
| 12:26.54 | ta[i]nted | ManxPower when i make some money i will paypal you some dough. |
| 12:27.03 | Darwin35 | yes |
| 12:27.05 | ta[i]nted | because you are the god damn ~man~. |
| 12:27.11 | *** join/#asterisk afe ([mBhymEq23@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 12:27.16 | Darwin35 | ztdummy is needed for timing interface |
| 12:27.16 | Eight | ManxPower: while you're being helpful: Any suggestions on how to fake a dialtone? I can use Playtones(dial) but I'm not sure how to make it stop as soon as the user starts dialing an extension... |
| 12:27.41 | puppet | WHy am I developing my own queue system? |
| 12:27.42 | ManxPower | Eight, You have to seriously think about why you want to simulate a dialtone. |
| 12:27.43 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 12:27.58 | ManxPower | puppet, The only reasons I can think of is that you are crazy. |
| 12:28.06 | Eight | ManxPower: I want to give the user a 'reassuring' dial tone for when they're making outgoing calls. |
| 12:28.11 | puppet | manxpower: or that i didnt get Park() to work? |
| 12:28.14 | Darwin35 | for after dial 9 for outside exten |
| 12:28.15 | puppet | manxpower: can be that ;p |
| 12:28.25 | Eight | Darwin35: exactly. |
| 12:28.34 | ManxPower | Eight, Asterisk and/or your VoIP device is SUPPOSED to do that already. |
| 12:28.49 | Darwin35 | no on sip calls |
| 12:28.56 | ManxPower | puppet, Since Park is for call parking, not queues, I can see why it's not working for you. |
| 12:28.58 | Darwin35 | you dial 9 and get deadair |
| 12:29.01 | ta[i]nted | Eight DISA()? |
| 12:29.23 | puppet | manxpower: Yeah But still im using queue for call parking now ;p |
| 12:29.27 | Eight | ManxPower: off hook, dial tone, dial extension 9, dial tone, dial number. |
| 12:29.31 | puppet | manxpower: but via webinterface is it gone be |
| 12:29.31 | Eight | ManxPower: that's what I"m trying to make. |
| 12:29.55 | ManxPower | Eight, And what about this is not working? I do that on all my systems. |
| 12:29.56 | ta[i]nted | puppet you are crazy |
| 12:30.09 | puppet | ta[i]nted: thanks ;p |
| 12:30.21 | Eight | ManxPower: when I use Playtones(dial), it doesn't stop as soon as the user enters the first digit of their extension. |
| 12:30.45 | ManxPower | Eight, I'm sorry, I can't help you set up Asterisk wrong. |
| 12:30.57 | Eight | ManxPower: I *know* I setup asterisk wrong =p |
| 12:31.03 | ta[i]nted | puppet u realize that http was never meant for connection state right? |
| 12:31.09 | Eight | ManxPower: That's why I'm asking questions =) |
| 12:31.13 | ManxPower | Eight, What device are you using to dial? |
| 12:31.15 | puppet | ta[i]nted: Hu? :) |
| 12:31.27 | ta[i]nted | puppet how will u refresh.. meta refresh, xmlhttp, or iframe? |
| 12:31.34 | puppet | ta[i]nted: check pm |
| 12:31.42 | Eight | ManxPower: I don't see how that matters... exactly. But a SIP phone. |
| 12:31.46 | ta[i]nted | yea i did |
| 12:31.58 | ta[i]nted | but wouldn't u want live status |
| 12:32.02 | ManxPower | Eight, SIP phones don't send a call to Asterisk until they think you are done dialing. |
| 12:32.10 | puppet | ta[i]nted: its live the moment you lost the page |
| 12:32.12 | ManxPower | Eight, now what BRAND of SIP phone are you using? |
| 12:32.17 | puppet | load* |
| 12:32.20 | Eight | ManxPower: I'm not talking about the initial dial. |
| 12:32.25 | Eight | ManxPower: I'm talking about once I'm already connected. |
| 12:32.51 | ManxPower | Eight, I STILL don't understand what you are doing. My users don't usually expect a dialtone after they are done dialing their number. |
| 12:32.59 | ta[i]nted | 1.0.6 coughed up this: Mar 11 04:32:38 WARNING[15433]: loader.c:440 load_modules: Loading module pbx_dundi.so failed! |
| 12:33.02 | Eight | They're NOT done dialing their number. |
| 12:33.07 | Eight | they're just hitting 9 to reach and 'outside line'. |
| 12:33.17 | ManxPower | Well there's your problem. |
| 12:33.19 | Eight | I want to fake a dial tone so they 'feel' like they have an actual outside line. |
| 12:33.33 | Eight | This is extremely common on phone systems I've used in the past, and I'd like to simulate it with Asterisk. |
| 12:33.51 | Darwin35 | like a nec or merlin system |
| 12:33.54 | ManxPower | 1) pick up the phone 2) hear dialtone 3) dial 9 4) hear dialtone 5) dial number. |
| 12:33.59 | Eight | Exactly. |
| 12:34.10 | Darwin35 | use ignorepat =9 might work also |
| 12:34.12 | Eight | [06:29] <Eight> ManxPower: off hook, dial tone, dial extension 9, dial tone, dial number. |
| 12:34.18 | ManxPower | Eight, That's how all my systems are set up. Nothink special. |
| 12:34.38 | ManxPower | Darwin35, ignorepat does not work for SIP. |
| 12:34.43 | Eight | ManxPower: So do you feel like sharing? Or did I just waste 10 minutes of both of our time? |
| 12:34.44 | Darwin35 | ahh ok |
| 12:35.13 | Darwin35 | so manx here is the issue as I see it |
| 12:35.27 | puppet | hmm why does the IAX link get /2 after it |
| 12:35.32 | puppet | it messes with me, I dont like that! ;D |
| 12:35.50 | Darwin35 | on a sip dial out plan you use _9NXX. ,1,Dial() |
| 12:36.01 | Darwin35 | but it does not drop back to a dial tone |
| 12:36.11 | Darwin35 | he wants it to drop back to a tone |
| 12:36.15 | ta[i]nted | in case anyone cares, remember to remove old libs from /usr/lib/asterisk/modules before upgrading asterisk |
| 12:36.30 | Darwin35 | then after they dial again for the tone to stop |
| 12:37.08 | Eight | ta[i]nted: what broke when you didn't? |
| 12:37.09 | ManxPower | Eight, Here is my dialplan on my SIPura: ([2-7]xxx|9,1[2-9]xx[2-9]xxxxxx|9,[2-9]xxxxxx) |
| 12:37.17 | ManxPower | the , means "contrinue dialtone" |
| 12:37.17 | ta[i]nted | Eight dundi |
| 12:37.20 | Darwin35 | he is usin exten => s,2,Background(Tonne) |
| 12:37.25 | Eight | ta[i]nted: ah, don't have that =) |
| 12:37.34 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
| 12:37.40 | Darwin35 | ahh ok |
| 12:37.41 | ManxPower | Now, if your SIP device is too braindead to support that, well you need more help than I can give. |
| 12:37.53 | Darwin35 | i did not know abotu the , |
| 12:37.56 | ta[i]nted | Eight what sip device are u using |
| 12:37.57 | Darwin35 | thats new |
| 12:38.04 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 12:38.20 | Eight | It doesn't have anything to do with the device. |
| 12:38.23 | ManxPower | Darwin35, Not at all. SIPura devices have supported "," since they first were relased. |
| 12:38.25 | Eight | I could be using dsp/console =p |
| 12:38.28 | ta[i]nted | lol |
| 12:38.36 | ManxPower | Eight, With SIP it's ALWAYS about the device. |
| 12:38.38 | Darwin35 | to me its new |
| 12:38.43 | Darwin35 | I have a GS 102 |
| 12:38.50 | Darwin35 | and have not tried that |
| 12:39.04 | ManxPower | Darwin35, A GS102 is not a SIPura. GS stuff is too braindead to continue dialtone |
| 12:39.08 | Eight | ManxPower: I give up. Have a nice day. |
| 12:39.15 | Darwin35 | ok |
| 12:39.33 | ta[i]nted | cool there is an eagi-spinx-test |
| 12:39.49 | ta[i]nted | now i can develop annoying voice to text recognition just like the big boys! |
| 12:39.52 | Darwin35 | I need the sphinx agi stuff |
| 12:39.54 | ta[i]nted | yay |
| 12:40.00 | Darwin35 | ta where did you find it |
| 12:40.04 | ManxPower | Darwin35, As I'm sure you know, SIP devices collect digits and then send the entire number to their server when they think the user is done dialing. So, as you can see, all dialplan stuff on a SIP device is configured on the SIP device. |
| 12:40.20 | Darwin35 | yes |
| 12:40.22 | ta[i]nted | in /var/spool/asterisk/agi-bin/eagi-spincter-test |
| 12:40.24 | Darwin35 | grrr |
| 12:40.40 | Darwin35 | thnks ta |
| 12:40.49 | ManxPower | Darwin35, That's why I make sure to only buy devices that support the SIP features I want. |
| 12:41.05 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 12:41.10 | Darwin35 | well back when I got the 102 it was all I could afford |
| 12:41.11 | RoyK | I just wonder |
| 12:41.15 | Darwin35 | things have changed |
| 12:41.23 | Darwin35 | I have a new job and some income now |
| 12:41.30 | RoyK | could we use sendfile() in asterisk to have the NIC offload the CRC? |
| 12:41.56 | ManxPower | Darwin35, You can sort of emulate things by using whatever "hotline" feature the device supports and then hack togather something using DISA, but it never works quite as you expect. |
| 12:42.01 | ta[i]nted | Darwin35 what do u do now if u don't me asking |
| 12:42.29 | Eight | Darwin35: http://lists.digium.com/pipermail/asterisk-users/2003-September/020147.html |
| 12:42.34 | Darwin35 | I developed a small pbx system aka soho with * |
| 12:42.38 | ta[i]nted | DISA sucks |
| 12:42.45 | Darwin35 | designed a small board |
| 12:42.45 | Eight | ~google asterisk fake dialtone |
| 12:44.03 | ta[i]nted | very unreliable for dialtone |
| 12:44.03 | Eight | ergh, too many links. |
| 12:44.03 | Eight | It's the first hit, anyways. |
| 12:44.04 | ManxPower | ta[i]nted, I know. |
| 12:44.04 | *** join/#asterisk Dirtysole (~Frank@cpe.atm0-0-0-1201091.0x503fc556.bynxx2.customer.tele.dk) |
| 12:44.18 | ta[i]nted | damn |
| 12:44.22 | puppet | Annoying that the uk forwarder doesnt send caller id |
| 12:44.23 | ta[i]nted | 1.0.6 still generating dead air |
| 12:44.58 | puppet | im getting alot of "flicker" |
| 12:45.02 | ta[i]nted | could certain providers block dialtone? is that possible? |
| 12:45.22 | ManxPower | ta[i]nted, Huh? |
| 12:45.22 | ta[i]nted | DISA() refuses to work on some cellular phones |
| 12:45.30 | Eight | ta[i]nted: really? how so? |
| 12:45.31 | Darwin35 | ok 1.0.6 is being a pain |
| 12:45.34 | Darwin35 | grrr |
| 12:45.43 | ManxPower | ta[i]nted, You tried Answer() before DISA? |
| 12:45.50 | ta[i]nted | yea |
| 12:46.00 | ta[i]nted | works fine on landlines |
| 12:46.09 | ta[i]nted | but only one particular cell carrier |
| 12:46.11 | ta[i]nted | forget which one |
| 12:46.12 | Eight | ta[i]nted: maybe you need to relax dtmf? |
| 12:46.16 | ta[i]nted | i think at&t |
| 12:46.47 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
| 12:47.19 | *** join/#asterisk IronIan (~IronIan@port-195-158-185-138.static.qsc.de) |
| 12:47.22 | *** join/#asterisk jeffik (~jeffik@user-v7kajt8.dialup.mindspring.com) |
| 12:47.35 | *** join/#asterisk rajo (~rajo@graphics.cs.uni-sb.de) |
| 12:47.56 | *** part/#asterisk IronIan (~IronIan@port-195-158-185-138.static.qsc.de) |
| 12:48.33 | Darwin35 | ok that fake dial tone works on the 102 ... |
| 12:48.37 | Darwin35 | thats a good fix |
| 12:49.35 | Eight | Darwin35: how'd you fake it exactly? |
| 12:49.47 | Eight | http://lists.digium.com/pipermail/asterisk-users/2003-September/020147.html this way? |
| 12:50.33 | Darwin35 | yes |
| 12:51.17 | Darwin35 | its fine |
| 12:51.52 | Eight | except there's no 'dialtone' file... |
| 12:52.15 | Darwin35 | I have one |
| 12:52.23 | Darwin35 | I think its in the addon sounds |
| 12:52.35 | Eight | I have those... I get a file not found though. |
| 12:52.52 | Darwin35 | let me find mine |
| 12:53.01 | puppet | http://www.unco.de/asterisk/dialtone.wav |
| 12:53.08 | Eight | puppet: 404 |
| 12:53.10 | puppet | there is one |
| 12:53.11 | Eight | Already tried =) |
| 12:53.11 | puppet | aha :/ |
| 12:53.15 | puppet | damn ;p |
| 12:53.24 | Essobi | Make sone. |
| 12:53.25 | Essobi | one. |
| 12:53.36 | Eight | It seems silly to play a wav, when it's just an easily generated tone. |
| 12:53.37 | Essobi | It is a DTMF. |
| 12:53.50 | Essobi | I thought that was the problem.. |
| 12:54.00 | puppet | http://www.telephonetribute.com/international.html |
| 12:54.06 | Essobi | Playtone does innterupt when digit collection starts. |
| 12:54.14 | puppet | http://www.nirvani.net/misc/telephone_audio/ |
| 12:54.15 | Eight | Essobi: well, I'd like to use Playtones(dial), except I can't figure out how to stop it after the first key press. |
| 12:54.24 | *** join/#asterisk telenieko (~telenieko@62-15-139-146.inversas.jazztel.es) |
| 12:54.28 | Essobi | WHY he isn't doing DTMF generation with his phone is beyond me thou. |
| 12:54.30 | telenieko | hi |
| 12:54.37 | puppet | http://asterisk.gnuinter.net/files/digium/phonecore/sounds/ |
| 12:54.38 | puppet | ;D |
| 12:54.43 | telenieko | Has anyone succeeded on settingup mplayer for music on hold ? |
| 12:54.46 | puppet | there are som eight :) |
| 12:54.55 | puppet | telenieko: why you wanne use mplayer? |
| 12:55.14 | telenieko | I want to connect and ASF stream from a local radio to my music on hold ;o) |
| 12:55.21 | Essobi | Eight ? |
| 12:55.34 | Eight | Essobi: what? |
| 12:55.39 | Essobi | WHY he isn't doing DTMF generation with his phone is beyond me thou. |
| 12:55.39 | Eight | Wiki: "Like Background, if Playtones is active when execution reaches the highest-priority command defined in for the extension, then the sound will continue to play until either the user dials an extension" |
| 12:55.42 | ta[i]nted | damn it broadvoice is frustrating |
| 12:55.48 | Eight | Essobi: who isn't? |
| 12:55.56 | Essobi | Eight Why are you not doing it with a dial plan? |
| 12:55.59 | Eight | ta[i]nted: ya it is... it works fine for me now that it's setup, but it's black magic getting setup the first time. |
| 12:56.09 | Essobi | Umm. A dialtone is dtmf too bub. :) |
| 12:56.23 | ta[i]nted | Eight no it was working just fine.. but recently there were some changes |
| 12:56.31 | Essobi | Why arn't you using the phone to provide the dialtone. |
| 12:56.34 | ta[i]nted | i can't place two simultaneous calls anymore |
| 12:56.38 | Eight | Essobi: 'The phone"? |
| 12:56.44 | *** join/#asterisk justinnnnnn (~justinnnn@c211-28-200-67.eburwd1.vic.optusnet.com.au) |
| 12:56.49 | Essobi | O, WTF are you playing a dialtone for? |
| 12:57.12 | Darwin35 | frick |
| 12:57.26 | justinnnnnn | im bored |
| 12:57.27 | Essobi | You have random payphones call you on PSTN and you provide dialtone for them? :) |
| 12:57.28 | Eight | Off Hook, Dial Tone, Dial 9, New Dial tone, Dial first digit-tone STOPS-continue dialing number |
| 12:57.29 | justinnnnnn | and i need money |
| 12:57.44 | Essobi | Eight Umm. what kind of phones. |
| 12:57.49 | Eight | Essobi: x-lite. |
| 12:57.58 | Essobi | Hah. |
| 12:58.02 | Essobi | Right then. |
| 12:58.03 | Eight | but it doesn't matter. |
| 12:58.11 | Eight | incidentally, it works fine with DISA for me. |
| 12:58.24 | Essobi | A few sip phones will do that with a dialtone. |
| 12:58.34 | *** join/#asterisk Dirtysole (~Frank@cpe.atm0-0-0-1201091.0x503fc556.bynxx2.customer.tele.dk) |
| 12:58.42 | Essobi | in a dialplan that is tftped |
| 12:58.54 | Eight | It, has, nothing, to do, with the phone. |
| 12:59.07 | Eight | I know what you're talking about... but that's NOT what I'm doing. |
| 12:59.07 | ta[i]nted | eight what device are you using |
| 12:59.25 | ta[i]nted | (just kidding) |
| 12:59.27 | Eight | I'm going to stop answering that question |
| 12:59.29 | Eight | hah. |
| 12:59.55 | ta[i]nted | u sure they need a reassuring dialtone? |
| 13:00.02 | ta[i]nted | why not give voice prompt |
| 13:00.05 | Essobi | Off Hook, Dial Tone, Dial 9, New Dial tone, Dial first digit-tone STOPS-continue dialing... that's has everything to do with a 7960 dialplan file. you define the 9, and the second dialtone in them.. |
| 13:00.09 | ta[i]nted | "this is a dialtone" |
| 13:00.16 | Essobi | hahaha |
| 13:00.25 | puppet | hmm aint there any way to get the API to merge to .gsm files? whil using mute |
| 13:00.30 | Essobi | "BOOP BOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOO..... |
| 13:00.33 | ta[i]nted | or just record a file where you say "duuuuuuuuuuuuu" |
| 13:00.37 | puppet | not mute, monitor |
| 13:00.51 | Eight | I think it's a bug with Playtones, to be honest. |
| 13:00.57 | ta[i]nted | Essobi lol.. that's an advanced dialtone |
| 13:00.58 | Eight | either that or it's missdocumented. |
| 13:01.05 | JunK-C | create a wav, then convert it to gsm? |
| 13:01.13 | JunK-C | ya can create gsm too with Monitor |
| 13:01.16 | Essobi | Background(Dialtone) would work. |
| 13:01.21 | Eight | Essobi: yes, we know. |
| 13:01.32 | Eight | Essobi: I just think it's silly to play a wave for a DTMF =p |
| 13:01.40 | ta[i]nted | Eight have you looked at the device at all? |
| 13:01.40 | Essobi | Yea, anyways.. I'm out. |
| 13:01.42 | Eight | or .gsm, or whatever. |
| 13:01.47 | Eight | ta[i]nted: go away =p |
| 13:01.54 | Eight | better yet, I will |
| 13:01.59 | ta[i]nted | sorry i'm just bitter that u have BV working |
| 13:02.00 | Essobi | I think it's just silly to you X-lite as a phone, but to each his own. :) |
| 13:02.17 | Essobi | ^use even |
| 13:02.18 | puppet | junk-c: i want it it to be one file and not to |
| 13:02.22 | puppet | junk-c: two* |
| 13:02.37 | Essobi | Anyways.. I'M OUT. |
| 13:02.44 | puppet | junk-c: THat means i want it to run sox infile outfile mixedfile |
| 13:03.00 | puppet | junk-c: ornot sox but right command :) |
| 13:03.02 | ManxPower | Essobi, These people are using devices that are too stupid to continue dialtone. |
| 13:03.04 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 13:03.28 | Darwin35 | grrr |
| 13:04.23 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
| 13:04.23 | *** join/#asterisk citats (~james@duff.gnuinter.net) [NETSPLIT VICTIM] |
| 13:04.23 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM] |
| 13:04.23 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) [NETSPLIT VICTIM] |
| 13:04.23 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) [NETSPLIT VICTIM] |
| 13:05.57 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 13:07.04 | puppet | junk-c: Test-in.wav Test-out.wav it gives thoose files. I want it to give Test.wav |
| 13:07.48 | ManxPower | puppet, the "m" option to monitor doesn't work? |
| 13:07.57 | puppet | manxpower: there is no m option |
| 13:08.03 | puppet | manxpower: im working threw manager API |
| 13:08.06 | [ro]nic3try | can anyone tell me how to set asterisk so that no RTP should cross by box ? |
| 13:08.20 | [ro]nic3try | pls.. |
| 13:08.21 | ManxPower | puppet, Perhaps you should look at "show applicaton record" |
| 13:08.21 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 13:08.37 | ManxPower | [ro]nic3try, "canreinvite=yes" |
| 13:08.39 | puppet | manxpower: Action: Monitor\n\rChannel: SIP/1001-b281\r\nFilename: Test |
| 13:08.49 | puppet | manxpower: no |
| 13:08.52 | *** join/#asterisk Jayden (~ircatjerr@65.170.43.34) |
| 13:08.54 | puppet | manxpower: thats not what i wanne do |
| 13:08.58 | ManxPower | [ro]nic3try, However, that won't work if you have "t" or "T" on the Dial line. |
| 13:09.03 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 13:09.06 | *** join/#asterisk mangaan89 (~peter@d515233A3.access.telenet.be) |
| 13:09.07 | puppet | manxpower: i wanne be able to have a Monitor button on my webinterface |
| 13:09.20 | puppet | manxpower: when I click monitor it starts to monitor that choosen channel |
| 13:09.37 | puppet | manxpower: But It seems like i have to mix it myself since the API lacks thoose functions |
| 13:09.39 | ManxPower | puppet, At the Asterisk CLI "show manager command monitor" |
| 13:09.55 | ManxPower | Notice the "Mix" option. |
| 13:09.58 | puppet | hmm |
| 13:09.58 | *** join/#asterisk IronIan (~IronIan@port-195-158-185-138.static.qsc.de) |
| 13:10.19 | puppet | stopid wiki ;p |
| 13:10.24 | puppet | think im gone create user and comment |
| 13:10.35 | puppet | manxpower: thanks ;D |
| 13:10.39 | ManxPower | puppet, Never trust the Wiki for something you can verify yourself. |
| 13:10.45 | mangaan89 | What are the mimnimum .conf files needed to run asterisk |
| 13:11.20 | ManxPower | mangaan89, /etc/asterisk/asterisk.conf I think |
| 13:11.27 | ManxPower | But it's pretty useless if you do that. |
| 13:11.33 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 13:12.06 | Darwin35 | [dial9] |
| 13:12.07 | Darwin35 | exten => 9,1,Goto(dialtone,s,1) |
| 13:12.07 | Darwin35 | [dialtone] |
| 13:12.07 | Darwin35 | exten => s,1,Answer |
| 13:12.07 | Darwin35 | exten => s,2,Background(dialtone) |
| 13:12.07 | Darwin35 | exten => s,1,Dial(${voip-provider}) |
| 13:12.20 | ManxPower | Darwin35, I take it you don't like DISA? |
| 13:12.30 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:12.31 | mangaan89 | Manxpower You have to be sure it is for a production enviroment ;-) |
| 13:12.34 | Darwin35 | I have not played with disa |
| 13:12.46 | Darwin35 | but disa gives to much control from what I read |
| 13:12.58 | Darwin35 | but I will try disa |
| 13:13.01 | Darwin35 | next |
| 13:13.04 | ManxPower | Darwin35, Your Dial will never be called in your example. |
| 13:13.05 | *** part/#asterisk doppiaemme (~max@131.175.126.186) |
| 13:13.30 | Darwin35 | ? |
| 13:13.46 | puppet | mmmmmmmmmmmmm SIP-1001-b281.mp3 |
| 13:13.53 | puppet | just got to set the variable now for filename ;D |
| 13:13.55 | puppet | manxpower: thanks |
| 13:13.56 | Darwin35 | needs /${exten} |
| 13:14.01 | ManxPower | Darwin35, Come to think of it, having two exten => s,1 lines is a much more serious problem with your example. |
| 13:14.19 | Darwin35 | I forgot to change it to 3 |
| 13:14.21 | IronIan | Hi everyone, the dev team is ordering me to ask my questions her. So for a start is anyone here with knowledge about coding a custom channel driver? |
| 13:14.37 | ManxPower | Darwin35, "background means play something and wait for digits" If you dial 1234 then Asterisk will call exten => 1234 |
| 13:15.43 | ManxPower | IronIan, A better place to ask is the asterisk-dev and asterisk-users mailing lists. |
| 13:15.47 | ManxPower | ~mailinglist |
| 13:15.48 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. |
| 13:16.05 | Darwin35 | yes so if it then goes to my area called voip-provider it only ha _011 and Nxx in it |
| 13:16.38 | mangaan89 | What are the minimum .conf files needed to run asterisk as a PSTN gateway |
| 13:17.06 | Eight | mangaan89: I seem to be getting away with extensions.conf and sip.conf |
| 13:17.07 | IronIan | MaxPower: asterisk-dev channel wants to ask me here: "<JunK-C> IronIan: goto #asterisk, that kind of question isnt for dev" |
| 13:17.15 | Eight | mangaan89: err, plus you'd need zap'. |
| 13:17.16 | ManxPower | mangaan89, Stop asking stupid questions. Set up Asterisk, then you'll understand what files you need. |
| 13:17.29 | Darwin35 | Dial should be Goto |
| 13:17.46 | ManxPower | IronIan, Only because your question is so general as to be useless. |
| 13:18.10 | ManxPower | IronIan, If you want to see how to write a channel driver, the only way is to look at an existing channel driver. |
| 13:18.13 | Darwin35 | Manx |
| 13:18.19 | mangaan89 | Manxpower Stupid questions?? You even could give me the answer :-)) |
| 13:18.22 | Eight | ManxPower is a very angry person. How sad. |
| 13:18.28 | Darwin35 | [dialtone] |
| 13:18.28 | Darwin35 | exten => s,1,Answer |
| 13:18.28 | Darwin35 | exten => s,2,Background(dialtone) |
| 13:18.28 | Darwin35 | exten => s,3,Goto(${voip-provider}) |
| 13:18.36 | puppet | darwin35: iik spam |
| 13:18.40 | puppet | ;D |
| 13:18.42 | puppet | brb |
| 13:18.46 | IronIan | ManxPower: My question in -dev was quite detailed. I just do not want to ask it here again, because its some lines longer and i think it needs a coder with knowledge in channel coding. |
| 13:18.48 | ManxPower | mangaan89, Yes, I could. I could spend a few hours researching your question and give you an answer. But that usually requires payment. |
| 13:18.48 | puppet | http://www.voip-info.org/tiki-index.php?page=Asterisk+Manager+API+Action+Monitor |
| 13:18.52 | puppet | manxpower: look ! ;D |
| 13:18.54 | IronIan | ManxPower: Sorry for that |
| 13:19.07 | ManxPower | IronIan, Then your only option is to really ask on the mailing lists. |
| 13:19.18 | mangaan89 | Thanks to Eight A try this !! |
| 13:19.19 | Eight | ManxPower: If you don't have anything nice to say, don't say anything at all. It's a simple rule that applies very well when 281 people are in the same channel =p |
| 13:19.19 | bjohnson | hahaha .. http://www.fuckinggoogleit.com/ |
| 13:19.44 | ManxPower | Darwin35, Try it. It won't work. |
| 13:20.34 | ManxPower | Darwin35, Leave out the goto, add an include => for the context you want the user to have access to. |
| 13:21.08 | mangaan89 | Manxpower I never ask you to do that ;-) |
| 13:21.22 | IronIan | ManxPower: its not a problem to write one, its a problem to narrow a problem in an existing one down and i need someone leading me to the asterisk side, cause i am really sure whats going on in the channel driver itself. I just need to know the parts in asterisks own code |
| 13:22.19 | ManxPower | IronIan, People that know those things tend not to hangout on #asterisk. They hang out on #asterisk-dev, but the nature of your question is really more suitable for THE MAILING LIST. |
| 13:22.42 | ManxPower | mangaan89, There is no documentaiton with the answer you want. |
| 13:22.50 | ManxPower | ~docs |
| 13:22.51 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:22.51 | [ro]nic3try | thx ManxPower |
| 13:22.54 | ManxPower | ~mailinglist |
| 13:22.55 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. |
| 13:22.55 | Darwin35 | ok got it |
| 13:23.40 | [ro]nic3try | ~weather LRBC |
| 13:23.44 | mangaan89 | Manxpower OK I try it myself let you now the answer !! |
| 13:23.53 | [ro]nic3try | ~weather LRBS |
| 13:23.55 | ManxPower | mangaan89, put it on the Wiki |
| 13:24.14 | ManxPower | [ro]nic3try, If you are going to play with the bot please do so in private and wash your hands after. |
| 13:24.25 | Darwin35 | [dialtone] |
| 13:24.25 | Darwin35 | exten => s,1,Answer |
| 13:24.25 | Darwin35 | exten => s,2,Background(dialtone) |
| 13:24.25 | Darwin35 | include => voip-provider |
| 13:24.26 | [ro]nic3try | ok.. sorry |
| 13:24.32 | Darwin35 | that worked |
| 13:24.38 | IronIan | ManxPower: Okay, okay..... i really understood that. But I don't like to get blamed from someone don't knowing my real intentions. |
| 13:24.54 | ManxPower | Darwin35, <dry tone> My advice frequently does. </dry tone> |
| 13:25.24 | Darwin35 | ? |
| 13:25.29 | bjohnson | wow .. Sixtel makes no warranties of any kind, written or implied, to the service in which it provides. |
| 13:25.33 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
| 13:25.33 | Darwin35 | drytone ? |
| 13:25.44 | bjohnson | I wonder if I can sell with those conditions |
| 13:25.45 | Darwin35 | ~dry tone |
| 13:25.47 | jbot | ACTION dries tone |
| 13:25.58 | ManxPower | Darwin35, Won't worry about it. |
| 13:26.03 | Darwin35 | ok |
| 13:26.34 | Darwin35 | wel this part is new to me . I am trying dif ways till I find a way I like it to lock my out going calls |
| 13:26.37 | bjohnson | maybe I'll resell sixtel service and make no warrantees that they'll honour the sale |
| 13:27.10 | Darwin35 | so I am listening and learning from you |
| 13:27.20 | *** part/#asterisk mangaan89 (~peter@d515233A3.access.telenet.be) |
| 13:27.22 | Darwin35 | I am your padawand |
| 13:27.28 | Darwin35 | you are my teacher |
| 13:27.31 | bjohnson | I really need to find a better Ontario DID provider |
| 13:27.40 | ManxPower | Darwin35, Can you be my sigar daddy? |
| 13:27.45 | ManxPower | Darwin35, Can you be my sugar daddy? |
| 13:27.55 | *** part/#asterisk IronIan (~IronIan@port-195-158-185-138.static.qsc.de) |
| 13:27.58 | Darwin35 | hehe |
| 13:28.16 | Darwin35 | wow this old laptop is a blast |
| 13:28.39 | Darwin35 | i just loaded a ibm think pad 380d with me |
| 13:28.56 | Darwin35 | its all it would take |
| 13:29.05 | Darwin35 | for righ now |
| 13:29.15 | Darwin35 | I will put some for linux lttr |
| 13:29.17 | ManxPower | bjohnson, Who do you use now? |
| 13:29.27 | Darwin35 | but needed a in bo for programming pay phones |
| 13:29.29 | *** join/#asterisk coldfeet (~cf@dsl-80-46-109-145.access.as9105.com) |
| 13:30.03 | ManxPower | I'll be in #asterisk-stable if people using 1.0.x have questions |
| 13:30.06 | *** part/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 13:31.47 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM] |
| 13:31.47 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) [NETSPLIT VICTIM] |
| 13:31.47 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) [NETSPLIT VICTIM] |
| 13:33.18 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 13:33.28 | fishboy1669 | hi guys hows finx |
| 13:33.39 | fishboy1669 | anyone having a good day? |
| 13:33.51 | fishboy1669 | im just thinking its friday so has to be good lol |
| 13:34.05 | fishboy1669 | still having problems with my customers system though |
| 13:34.32 | fishboy1669 | for some reason the phones seem to randomly get upset |
| 13:34.38 | fishboy1669 | ip2006 telcom |
| 13:34.44 | Darwin35 | jackpot |
| 13:34.56 | fishboy1669 | wird thing is that if i ping them from the server it seems to comeback to life |
| 13:35.08 | fishboy1669 | yay darwin sounds like u are haing fun |
| 13:36.50 | ta[i]nted | any AGI gurus? |
| 13:37.33 | Darwin35 | I just got the call I have been waiting on |
| 13:37.46 | Darwin35 | the wicked wich is dead |
| 13:37.50 | Darwin35 | witch |
| 13:38.02 | Darwin35 | <PROTECTED> |
| 13:38.18 | ta[i]nted | lol |
| 13:38.22 | ta[i]nted | translation |
| 13:38.24 | ta[i]nted | ? |
| 13:38.56 | jakepdev | ehat do you want to know about AGI? |
| 13:39.27 | *** join/#asterisk vcpalm (~msn@cc59163-a.ensch1.ov.home.nl) |
| 13:39.28 | fishboy1669 | darwin u getting a devorce? |
| 13:39.29 | fishboy1669 | lol |
| 13:41.21 | *** join/#asterisk dcornewell (~dcornewel@proxy5.med-web.com) |
| 13:41.44 | *** join/#asterisk dubphil (~dubphil@80.124.137.201) |
| 13:41.53 | jakepdev | tainted - what do you want to know about AGI? |
| 13:42.29 | ta[i]nted | jakepdev how do i execute DIal(SIP/12345@provider.com) in AGI? |
| 13:43.05 | ta[i]nted | $objAGI->exec("DIAL SIP/12345@provider.com"); # doesn't work |
| 13:44.56 | jakepdev | have you looked here? http://home.cogeco.ca/~camstuff/agi.html |
| 13:45.07 | *** join/#asterisk jmacz (~jmacz@201.245.167.104) |
| 13:45.41 | ta[i]nted | yea |
| 13:45.57 | ta[i]nted | $objAGI->exec("DIAL SIP/provider.com/12345"); # works |
| 13:46.12 | ta[i]nted | but i was wondering how i could send the other syntax |
| 13:46.17 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM] |
| 13:46.17 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) [NETSPLIT VICTIM] |
| 13:46.17 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) [NETSPLIT VICTIM] |
| 13:46.43 | jakepdev | your answer lies here - http://lists.digium.com/pipermail/asterisk-users/2004-April/043110.html |
| 13:47.39 | *** join/#asterisk backfired (~asterisk@217.159.105.46) |
| 13:48.51 | ta[i]nted | $objAGI->exec("DIAL SIP/12345/@provider.com"); # like this? |
| 13:48.58 | backfired | Hi there, I am new to this channell, can anybody please help me with a problem of connecting Asterisk to a VegaStream 50 BRI Gateway? |
| 13:49.00 | jakepdev | that looks right |
| 13:49.14 | jakepdev | oops |
| 13:49.18 | jakepdev | backslash is \ |
| 13:49.45 | lordcian | can anyone point me to a good example small business dialplan? I am understanding more and more of the extensions.conf options, but I'm having a problem seeing the 'big picture' |
| 13:50.07 | jakepdev | lordcian - why not just use the web interface |
| 13:50.25 | lordcian | which one do you like? AM? |
| 13:50.35 | puppet | lordcian: AM? |
| 13:50.42 | lordcian | Asterisk Manager? |
| 13:50.58 | jakepdev | AMP? |
| 13:51.10 | jakepdev | AMP seems to work well |
| 13:51.17 | puppet | asjoyner: Im building my own but its purley for call handling, monitoring, redirecting etc |
| 13:51.23 | puppet | jakepdev: amp? ;P |
| 13:51.38 | dubphil | I'm back, so my question was what hardware to invest in order to use about 200 phones (not analog) with asterisk in my office ? |
| 13:51.43 | lordcian | i've been trying to get AstMan to work: astman.sourceforge.com, because it seemed easiest, but i'm having a perl compliation issue... |
| 13:51.47 | jakepdev | http://sourceforge.net/projects/amportal/ |
| 13:52.13 | julianjm | Hello, what's the best way to receive faxes with an asterisk pbx? I mean, the zaptel tdm drivers are broken from what i've read in the mailing list... If I deploy an asterisk instalation in a small office (1BRI, 10extensions), what hardware do I need? |
| 13:52.18 | jakepdev | it comes loaded with *@home |
| 13:53.03 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 13:53.10 | Darwin35 | chan_zap.c: In function `pri_dchannel': |
| 13:53.10 | Darwin35 | chan_zap.c:9063: error: structure has no member named `forwardnum' |
| 13:53.10 | Darwin35 | gmake[1]: *** [chan_zap.o] Error 1 |
| 13:53.35 | puppet | hmm my mpg123 is strange |
| 13:53.40 | Darwin35 | anyone else getting this with 1.0.6 |
| 13:55.33 | Darwin35 | man madplay plays so much nicer th mpg123 |
| 13:56.31 | Darwin35 | this error is killing me |
| 13:56.35 | Darwin35 | chan_zap.c: In function `pri_dchannel': |
| 13:56.35 | Darwin35 | chan_zap.c:9063: error: structure has no member named `forwardnum' |
| 13:56.35 | Darwin35 | gmake[1]: *** [chan_zap.o] Error 1 |
| 13:57.45 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) [NETSPLIT VICTIM] |
| 13:58.06 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) [NETSPLIT VICTIM] |
| 14:00.08 | *** join/#asterisk backfired (~asterisk@217.159.105.46) [NETSPLIT VICTIM] |
| 14:00.38 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-1-2-120.w81-249.abo.wanadoo.fr) |
| 14:00.53 | backfired | Hello again, I was wondering whether anyone had had any experience with setting up Asterisk with a VegaStream 50 box? |
| 14:01.00 | jhiver | hi everybody |
| 14:01.12 | Darwin35 | vegastream never heard of it |
| 14:01.15 | dcornewell | Hi, is anyone here familiar with the asterisk source code? Particularly res_agi.c I want to add a RECORDEX FILE command |
| 14:01.16 | jhiver | don't know this hardware... |
| 14:01.26 | jhiver | is this a SIP phone of some kind? |
| 14:01.52 | backfired | it is a BRI ISDN SIP/H232 gateway |
| 14:02.00 | *** join/#asterisk mrgoby (~mrgoby@exchange.synergybroadband.com) |
| 14:02.12 | Darwin35 | dcorn add the code diff it againsed the src and then send a patch to the digium list explaining the change |
| 14:02.27 | jhiver | hah I have similar kind of equipement here and it's a complete mess to configure... |
| 14:02.35 | jhiver | I can't get my head around this |
| 14:02.52 | dcornewell | Darwin - well, I was hoping to talk to someone about the inner workings before I go tearing into it. not my code, you know |
| 14:02.55 | Darwin35 | well unless your rubber getting your head around things is hard |
| 14:03.12 | Darwin35 | the code is open src |
| 14:03.23 | Darwin35 | that means add fix change |
| 14:03.29 | jhiver | Quescom AS/400 GSM VoIP Gateway... anybody ever work with this stuff? |
| 14:03.29 | Darwin35 | then send to the list |
| 14:03.37 | backfired | I cannot seem to get asterisk to register to it. Was just wondering if anyone had an example. |
| 14:03.44 | mrgoby | i'm having some issues with the manager interface. it is transfering my call to the context i want, and the extension i want, but it is going the priority i specify minus one, and it isnt actually executing that command, just hanging, even though "show channels" from the CLI shows it is on that extension and priority |
| 14:03.50 | RoyK | jhiver: AS/400? |
| 14:03.58 | jhiver | yeah |
| 14:04.02 | mrgoby | using redirect, that is |
| 14:04.11 | jhiver | ?? |
| 14:04.19 | jhiver | don't understand |
| 14:04.34 | *** join/#asterisk Mike9 (~sturdee@sun.mikesweb.com) |
| 14:05.10 | jhiver | It's this family of products: |
| 14:05.12 | jhiver | http://www.quescom.com/quescom400_index.asp |
| 14:05.38 | jhiver | I'm sure I'll get my head around it sometime... it's just a bitch to work with |
| 14:06.02 | jhiver | especially the windows nt console... biew :( |
| 14:06.04 | Mike9 | quick comparison... what kind of lateney is asterisk showing to cisco 7940/7960 phones in 'sip show peers' for some? |
| 14:06.08 | dubphil | jhiver: what the prob I've got one |
| 14:06.49 | RoyK | jhiver: er. I thought about the AS/400, as in IBM Advanced System 400 |
| 14:07.07 | RoyK | s/advanced/administrative/ |
| 14:07.09 | RoyK | or something |
| 14:08.37 | puppet | mrgoby: i got same problem |
| 14:08.45 | puppet | mrgoby: send it to extention 2 and its ok ;p |
| 14:09.05 | puppet | mrgoby: havent coded the interface for redirect yet ill see how ill solve it then |
| 14:10.57 | mrgoby | i cant get it to connect no matter what.... i guess I'll just use a hack-around of a context with only one extension that is a wildcard and has a goto to the real context and extension i want.... though tihis is ugly |
| 14:13.37 | ta[i]nted | I GOT IT TO WORK!!!!!!!!!!!!!!!! |
| 14:13.38 | puppet | mrgoby: can u post your eextension file on pastebin? and the piece of code u send? |
| 14:14.05 | *** part/#asterisk vladuk (~vk@mustang.bzcrew.com) |
| 14:14.09 | ta[i]nted | GOD DAMN M' F'IN YEA BITCH YEA!!!!!!!!!1! |
| 14:14.16 | puppet | ta[i]nted: what? |
| 14:14.17 | mrgoby | extension looks like this: |
| 14:14.25 | mrgoby | [default] |
| 14:14.26 | puppet | mrgoby: paste on pastebin.ca :) |
| 14:14.29 | sambal | ~pastebin |
| 14:14.31 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 14:14.34 | puppet | dont spam channel please :) |
| 14:14.40 | ta[i]nted | i got asterisk to call my desk phone |
| 14:15.12 | ta[i]nted | j/k |
| 14:15.17 | puppet | hmmpf |
| 14:16.02 | puppet | its evul |
| 14:16.35 | ta[i]nted | i <3 * again |
| 14:16.59 | puppet | WHen using mix: 1 in the api call |
| 14:17.03 | puppet | it ignores filename :s |
| 14:17.16 | ta[i]nted | which mix |
| 14:17.17 | ta[i]nted | soxmix |
| 14:17.20 | ta[i]nted | ? |
| 14:17.26 | puppet | main problem is * |
| 14:17.33 | puppet | gone show you |
| 14:17.47 | mrgoby | http://pastebin.ca/7245 |
| 14:17.59 | Katty | morning |
| 14:18.35 | mrgoby | no matter what i do, it just hangs, then the t exten activates |
| 14:18.45 | mrgoby | all indications from asterisk are that it works |
| 14:19.28 | puppet | http://pastebin.ca/7246 < Filename is ignored :s |
| 14:20.07 | puppet | mrgoby: try add wait |
| 14:20.36 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E179.versanet.de) |
| 14:20.39 | Nebukadneza | hi ... |
| 14:20.40 | AgiNamu | ? |
| 14:20.58 | mrgoby | i dont understand |
| 14:21.34 | puppet | mrgoby: http://pastebin.ca/7247 |
| 14:21.46 | Nebukadneza | i got a more or less big problem with iaxclient (the library) and iaxcomm .... i compiled the lib and everything was okay - testcall compiled too. but when i try to compile iaxcomm i get the following errors: http://rafb.net/paste/results/JctBOB30.html |
| 14:22.04 | puppet | ta[i]nted: understand what I mean now? |
| 14:22.09 | Nebukadneza | (more of those errors ... later some with wx stuff) |
| 14:22.40 | mrgoby | i've tried 2, 3, 4, everything.... it always goes to priority -1 and it alwasy doesnt connect |
| 14:22.48 | puppet | mrgoby: i added more |
| 14:22.58 | puppet | mrgoby: s,2,Goto(s,1) |
| 14:23.02 | puppet | mrgoby: and then changed some other |
| 14:23.09 | mrgoby | is this due to the async_goto in manager.c , you think ? |
| 14:23.25 | *** part/#asterisk backfired (~asterisk@217.159.105.46) |
| 14:23.29 | *** join/#asterisk seong (~seong@218.208.204.115) |
| 14:23.36 | puppet | mrgoby: no clue but try that i wrote im using something like that |
| 14:23.45 | *** part/#asterisk stustu (~stustu@fluffy.fatburen.org) |
| 14:23.46 | mrgoby | right, the problem is that it will never connect to any of them, exept on timeout or input |
| 14:24.04 | mrgoby | s/exept/except/g |
| 14:24.04 | puppet | mrgoby: try it ;p |
| 14:24.13 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) |
| 14:24.16 | mrgoby | i have, it doesnt work |
| 14:24.20 | mrgoby | :0( |
| 14:24.27 | puppet | aha :( |
| 14:24.32 | puppet | can you show me the log? |
| 14:25.11 | mrgoby | unfortunately i'm at work and dont have access to the machine right now |
| 14:25.22 | puppet | dont you have ssh access? |
| 14:25.25 | Essobi | Tehe |
| 14:25.33 | mrgoby | no, it is at school in the lab |
| 14:25.36 | mrgoby | :-) |
| 14:25.39 | puppet | aha |
| 14:25.47 | mrgoby | no control of firewall |
| 14:25.51 | puppet | hard to test then :/ |
| 14:25.57 | mrgoby | though iax calls work fine going outbound :-D |
| 14:26.05 | puppet | ;p |
| 14:26.18 | puppet | but how do you edit the config? |
| 14:26.19 | Essobi | I wrote a XMLHttpRequest page for callerID. |
| 14:26.48 | Essobi | That backends into out local CM database. |
| 14:26.49 | puppet | essobi: help me with my problem instead ;D |
| 14:26.52 | *** join/#asterisk ke4qqq (~savirc@static-cb-68-115-212-155.spa.sc.charter.com) |
| 14:26.52 | Essobi | PSSH. |
| 14:27.01 | Essobi | JOO GO NOW! |
| 14:27.08 | Essobi | I'm the * nazi. |
| 14:27.12 | puppet | essobi: http://pastebin.ca/7246 |
| 14:27.15 | Essobi | NO HELP FOR YEW |
| 14:27.16 | puppet | essobi: goot then fix that ;D |
| 14:27.44 | Essobi | you're luck I don't have any open tickets. |
| 14:27.47 | Essobi | :P |
| 14:27.51 | *** join/#asterisk Xinhua123 (foobar@i-195-137-6-228.freedom2surf.net) |
| 14:27.52 | puppet | :p |
| 14:27.58 | puppet | LOL IM STUPID |
| 14:28.00 | puppet | HIT ME |
| 14:28.08 | AgiNamu | puppet, no I won't hit you. |
| 14:28.11 | Essobi | So um... Wha's the problem? |
| 14:28.17 | puppet | Its FILE! Not Filename! |
| 14:28.18 | puppet | lol |
| 14:28.19 | AgiNamu | I will tell you to logon to your colo'd machine, and type "route delete default" |
| 14:28.21 | AgiNamu | as root |
| 14:28.41 | Essobi | What what? You dont't want an Mp3? |
| 14:29.13 | Essobi | What's the problem? |
| 14:29.19 | Essobi | AgiNamu Maha. |
| 14:30.12 | puppet | Testing-dsadadadasda3-3434.mp3 |
| 14:30.16 | puppet | there |
| 14:30.23 | puppet | essobi: nah it was named standard name before |
| 14:30.26 | puppet | SIP-1001-b9dd.mp3 |
| 14:30.30 | puppet | But I gave it wrong paramater |
| 14:30.37 | puppet | Filename instead for just File |
| 14:30.55 | Essobi | AgiNamu echo "\:rf:rm" | awk -F '{print echo $3" "$2" "$1 }' | sh |
| 14:31.16 | puppet | I hate writing API to an API! ;P |
| 14:31.19 | Essobi | puppet.. Use the record command in your dialplan.. you can name it there. |
| 14:31.29 | puppet | essobi: nah not what Im after |
| 14:31.33 | puppet | essobi: for my webinterface |
| 14:31.51 | Essobi | So umm.. use the record function. |
| 14:31.54 | puppet | essobi: seeing active connections, aha nice, someone is talking in phone "Monitor" |
| 14:32.00 | puppet | essobi: poff monitors that line |
| 14:32.21 | AgiNamu | So, anyone have any details on the SGI Asterisk machines? |
| 14:32.34 | AgiNamu | I know they're boasting ungodly performance |
| 14:33.10 | AgiNamu | SGI might finally provide the port density Asterisk has been lacking |
| 14:33.28 | *** join/#asterisk dcornewell (~dcornewel@proxy5.med-web.com) |
| 14:33.30 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 14:33.54 | ariel_ | Hello everyone |
| 14:34.02 | AgiNamu | Hello, ariel_, welcome to #asterisk. |
| 14:34.33 | AgiNamu | heh, SGI should buy Digium! that'd be fun |
| 14:34.36 | johnnyb | Does anyone know how to use sox to encode .wav files that asterisk can use? The GSM files it produces are of bad quality, but I can't seem to get a wav format that asterisk will accept. |
| 14:34.49 | Essobi | AgiNamu Bite your tongue. |
| 14:34.57 | AgiNamu | Johnnyb, have you googled the wiki? |
| 14:35.02 | AgiNamu | Essobi why? |
| 14:35.08 | AgiNamu | SGI kicks ass in hardware |
| 14:35.12 | puppet | essobi: understand what i want with it now? :) |
| 14:35.34 | AgiNamu | If anyone, I'm sure they can figure out how to put 2000+ ports on a single machine |
| 14:35.37 | Essobi | pupper you're using the monitor port? |
| 14:35.49 | puppet | essobi: works fine now canshow you when its done :) |
| 14:35.54 | AgiNamu | instead of this "no more than a few cards cause there's conflicts" |
| 14:36.00 | johnnyb | AgiNamu: Yes, and it has great information on converting wav to gsm, but they have terrible quality. |
| 14:36.07 | Essobi | puppet Sure. |
| 14:36.15 | AgiNamu | johnnyb, all my GSM files sounds as good as gsm |
| 14:36.16 | Essobi | What's it wrote in? |
| 14:36.20 | lordcian | I have a question re: installation of AMP: it expects /var/www as doc root. help with this? |
| 14:36.26 | Essobi | Johny sox $name -r 8000 -c1 `echo $name | cut -f1 -d"."`.gsm resample -ql |
| 14:36.28 | AgiNamu | maybe you have your source wav screwed? |
| 14:36.31 | Essobi | Err no.. |
| 14:36.39 | johnnyb | Essobi: yep, that gets gsm files. |
| 14:36.54 | Essobi | Johny sox whatever.wav -r 8000 -c1 whatever.gsm resample -ql |
| 14:37.07 | ariel_ | lordcian, there is a an area that might help more with amp. it's at #amportal |
| 14:37.09 | Essobi | Works great for me. |
| 14:37.11 | johnnyb | Essobi: That produces terrible GSM files. |
| 14:37.19 | Essobi | Is your .wav file shit to begin with? |
| 14:37.20 | lordcian | thanks, joining |
| 14:37.20 | Essobi | :) |
| 14:37.35 | Essobi | amp has their own channel? |
| 14:37.43 | johnnyb | Essobi: no, it's great. It sounds wonderful, but it uses a pretty full dynamic range, which is why GSM is inappropriate. |
| 14:37.43 | Essobi | Haha.. I might to harass them. |
| 14:38.02 | *** join/#asterisk d1zzy (~dizzy@developer.user) |
| 14:38.03 | johnnyb | It's not going to be encoded anyway -- it's for the answering message from the ZAP lines. |
| 14:38.09 | Essobi | So play the wav and eat up your CPU. |
| 14:38.10 | Essobi | :) |
| 14:38.24 | johnnyb | Essobi: I'm trying, but asterisk won't recognize it. |
| 14:38.28 | Essobi | WAV to U/Alaw.. |
| 14:38.33 | Essobi | Hmm.. Convert it to a |
| 14:38.33 | Eight | man, BV is so fragile... I have no idea what I might have done to break it now. |
| 14:38.35 | Essobi | PCM? |
| 14:38.43 | Essobi | BV? |
| 14:38.48 | Eight | Essobi: Broadvoice. |
| 14:38.51 | Essobi | Ah. |
| 14:38.52 | Essobi | Nice. |
| 14:39.04 | Eight | I can still make outgoing calls, but not incoming =p |
| 14:39.07 | johnnyb | Essobi: Tried A-law, U-law, and PCM. Also, it should give less CPU time, bercause asterisk doesn't have to decode it. |
| 14:39.24 | d1zzy | I have configured asterisk to register to a SIP voip provider. it registers fine. I can call through the provider fine. if someone calls my number from the provider, immediatly he gets a "not available" tone. also tcpdump shows that I get an INVITE and I reply with Not Found. what could be the possible cause ? |
| 14:39.25 | Eight | And I appear to register fine. |
| 14:39.27 | *** join/#asterisk gst (~gst@wireless.sysfrog.org) |
| 14:39.31 | Essobi | I've got 4 propietary softswitches, I'm using * asterisk to call route to and fro now. :) |
| 14:39.37 | Eight | d1zzy: hey, that sounds familiar. |
| 14:39.45 | Essobi | johhny Yup.. correct assumption. |
| 14:39.52 | d1zzy | from the headers seems they use Cisco-SIPGateway/IOS-12.x |
| 14:39.53 | johnnyb | I've tried sox input.wav -c 1 -b -U output.wav, but that gives an unrecognizable file by asterisk. |
| 14:39.54 | Essobi | Hmm. Sounds like something is shifty with the source wav. |
| 14:40.04 | Essobi | Use somethng other then sox. |
| 14:40.08 | Essobi | Lemme see... |
| 14:40.13 | Eight | d1zzy: I'm having the exact same issue with Broadvoice right now. |
| 14:40.19 | johnnyb | Is there a specific bitrate that I need to encode to? |
| 14:40.29 | d1zzy | Eight: and you have any leads on this ? |
| 14:40.30 | Darwin35 | app_rx and tx fax is broken due to a lrent statement |
| 14:40.42 | Essobi | Darwin35 So fix it. |
| 14:40.48 | [ro]nic3try | rtp.c 298: processrfc3389:RFC3389 support incomplete ... why ? |
| 14:40.49 | Eight | d1zzy: nope =/ |
| 14:40.55 | Essobi | johnnyb I dunno. |
| 14:41.05 | puppet | afk coding webmanagment |
| 14:41.06 | Essobi | nice That's comfort noise isn't it? |
| 14:41.22 | Eight | d1zzy: Just been fiddling with stuff, then decided to try my new menu setup from a landline, and all of a sudden I can't get into asterisk from PSTN via broadvoice =/ |
| 14:41.34 | d1zzy | the address in the "To:" header of the SIP INVITE request has to match the local host ? it seems to be some other IP in there, is my number@someotherIP |
| 14:41.35 | Eight | whoa, never tcpdump without args from an SSH session =) |
| 14:41.47 | [ro]nic3try | rtp.c 298: processrfc3389:RFC3389 support incomplete ... what could be the problem ? |
| 14:41.51 | Essobi | yea.. RTP comfort noise.. Just ignore it nic3try. |
| 14:41.53 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 14:41.55 | Essobi | Disable it if you can. |
| 14:42.04 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 14:42.07 | [ro]nic3try | ok thx |
| 14:42.17 | Essobi | It's comfort noise.. doesn't do didly for you. |
| 14:42.24 | Essobi | http://www.faqs.org/rfcs/rfc3389.html |
| 14:43.23 | Essobi | It's part of VAD. |
| 14:44.03 | Essobi | Eight Haha. :) |
| 14:44.35 | Essobi | Alas his terminal finds woes in an vicious cycle of SSH and tcpdump. |
| 14:44.48 | Eight | Ya, so much for my scroll buffer =p |
| 14:44.48 | ariel_ | [ro]nic3try, that is when it stops sending the stream when there is no sound. Asterisk does not support this due to it's needs to stay in the stream. So you need to turn of silence suppression. |
| 14:45.17 | Eight | the machines are on a LAN too, so it got off to a pretty quick start. |
| 14:45.20 | Nugget | don't cross the streams! |
| 14:45.26 | Eight | Nugget: LOL! |
| 14:45.49 | [ro]nic3try | i'm try for the last 6 hours to make RTP to skip Asterisk when calling to a SER, but no chance ... does anyone know the condition for making that possible.. pls |
| 14:46.05 | Alexis | Is someone using a BT100 from grandstream here ? |
| 14:46.08 | gst | hmm.. is chan_h323 supposed to work? on debian it seems to be broken & when using the xorcom debs (including openh323 dependencies) it partly works but i can't hear anything (on both sides). |
| 14:46.17 | gst | just silence |
| 14:46.19 | Alexis | i can't make a transfert |
| 14:46.49 | gst | but the debug log says that the connections is established - and when doing a playback it is started too (according to the log) |
| 14:46.56 | ariel_ | gst, follow the readme to the point. h323 is a bear to configure and to make work. That is why most here don't use it. |
| 14:47.44 | *** join/#asterisk cbachman (~chatzilla@129.105.7.250) |
| 14:48.24 | ariel_ | [ro]nic3try, if your sip devices are using the same codec then you can do this with canreinvite=yes But if there using different codec's asterisk needs to be in the stream of things. |
| 14:49.09 | [ro]nic3try | it's the same codec |
| 14:49.12 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 14:49.22 | gst | ariel_: ok - i hoped that the xorcom packages would work. the strange thing is that everything seems to be normal, no warnings - no segfaults - just silence. i'll try to compile it myself - does chan_oh323 work better? |
| 14:49.29 | [ro]nic3try | i use canreinvite.. but no chance |
| 14:50.01 | ariel_ | gst, I don't use h323 nor oh323 its too much trouble to get it working in my book |
| 14:50.53 | fotang | I dial a user on asterisk on the LAN, it rings, he takes the call, but neither of us can hear the other. dead silence. any idea what we're doin wrong? |
| 14:51.09 | johnnyb | Okay, I figured it out -- the official way of encoding for asterisk: |
| 14:51.13 | johnnyb | sox ok.wav -r 8000 -c 1 -s ok2.wav resample -ql |
| 14:51.39 | johnnyb | MUCH MUCH better dynamic range without clicking. |
| 14:51.49 | johnnyb | Even on VoIP phones. |
| 14:51.50 | ariel_ | fotang, more info is needed. Like if your behind a nat and also what devices are you using? |
| 14:52.17 | fotang | oh sorry, more info: |
| 14:52.37 | ariel_ | [ro]nic3try, if the devices are behind a nat server it might not be able to hand off the call. |
| 14:53.22 | [ro]nic3try | none of the device is behind a nat |
| 14:53.44 | *** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
| 14:53.46 | fotang | asterisk is runnin on SuSE linux, on a NAT box but were in the internal interface, we are both using X-lite softphone on XP (atleast learned somethin from /.) |
| 14:54.02 | [ro]nic3try | the strange thing is that , when a conect to another SER server.. it works |
| 14:54.04 | fotang | s/were/we're |
| 14:54.31 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 14:54.31 | fotang | voive mail voices perfectly, only live talkin doesnt work: total silence |
| 14:54.48 | ariel_ | fotang, make sure you have nat=no since the devices are not behind a nat but on it's own network |
| 14:54.50 | fotang | voice mail works, i mean |
| 14:55.00 | fotang | ok |
| 14:55.42 | JunK-Y | <PROTECTED> |
| 14:55.42 | JunK-Y | Mar 11 09:53:35 NOTICE[3184]: app_dial.c:884 dial_exec_full: Unable to create channel of type 'Zap' (cause 0) |
| 14:55.54 | JunK-Y | wtf damn it, ive reloaded my kernel modules |
| 14:56.03 | [ro]nic3try | ariel_: when i put a call to ser.. the invite has one ip address and the rtp streams has other ip |
| 14:56.06 | JunK-Y | and i never touched that machine in a week |
| 14:56.11 | nestAr | no alarms on the circuit JunK-Y? |
| 14:56.22 | nestAr | that's the kind of error i get if my PRI is down |
| 14:56.53 | JunK-Y | damn, my patch for status isnt on that machine |
| 14:56.55 | JunK-Y | god man |
| 14:58.12 | *** join/#asterisk tessier (~treed@210.245.97.233) |
| 14:58.16 | ariel_ | [ro]nic3try, I don't use ser so I don't know why but sounds like there is sometype of nat involved |
| 14:58.41 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4463:7269:1:20e:a6ff:fe66:c5a3) |
| 14:58.47 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 14:58.47 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:00.06 | [ro]nic3try | ariel_ ok.. thx anyway |
| 15:00.35 | ariel_ | fotang, do you have the following in the sip.conf general section: bindaddr=0.0.0.0 externip = 68.142.20.XXX |
| 15:00.35 | ariel_ | localnet = 192.168.0.0/255.255.255.0 |
| 15:04.16 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 15:07.29 | mrgoby | i found this line in ast_async_goto(), which explains why i am put in priority-1 on redirect.... dont understand why it is this way though |
| 15:07.48 | mrgoby | chan->priority = priority - 1; |
| 15:07.50 | d1zzy | Eight: hey I have solved my problem, I was using register => ......@host/<extension> |
| 15:08.02 | d1zzy | and asterisk didnt found <extension> in the context of that SIP "friend" |
| 15:08.06 | mrgoby | not sure if this is intentional |
| 15:08.12 | Jearil | has anyone gotten a Cisco 7960 phone to work with asterisk without buying a support contract? |
| 15:08.16 | d1zzy | so I removed the /<extension> at all to go to the default entry of that context :) |
| 15:09.01 | Eight | d1zzy: I'm still not sure what my problem was, but I fixed it. |
| 15:09.01 | JunK-Y | nestAr: was the remote end ya. |
| 15:09.07 | JunK-Y | nestAr: thx for tips |
| 15:09.15 | nestAr | JunK-Y: np |
| 15:09.16 | nestAr | ;) |
| 15:09.26 | Eight | d1zzy: honestly, the leading possibility is that ordering of values in [sip.broadvoice.com] in sip.conf is relevant, which boggles the mind. |
| 15:09.32 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 15:09.47 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 15:09.52 | sambal | Jearil: you need a SIP license |
| 15:09.53 | riksta | Jearil: you need the sip firmware |
| 15:09.57 | d1zzy | Essobi: so that RFC3389 comes from the client... I dont understand how can I disable that from GnomeMeeting :) (I am using it with oh323 to connect to asterisk) |
| 15:10.03 | sambal | or image ;) |
| 15:10.06 | riksta | sambal: :) |
| 15:10.16 | Eight | d1zzy: I don't have the patience right now to go through all the permutations to figure out what breaks. |
| 15:10.29 | sambal | oh and you can't upgrade to the latest sip version in one step |
| 15:10.36 | sambal | you have to start at version 3 or 4 |
| 15:10.46 | Jearil | lovely.. and I assume the only way to get the firmware is to buy a license from cisco? |
| 15:11.04 | fotang | [15:54]#asterisk ariel_:fotang, make sure you have nat=no // where do i put that? /etc/skinny.conf, right? that's the only nat= i can find |
| 15:11.36 | sambal | that's a way yes |
| 15:11.37 | jontow | hmm.. i'm getting an error when i try to use STREAM FILE via AGI : |
| 15:11.39 | jontow | Mar 11 10:04:03 WARNING[557]: file.c:1073 ast_waitstream_full: Wait failed (Interrupted system call) |
| 15:11.44 | jontow | anyone seen it / know how to fix it? |
| 15:11.45 | sambal | officially you have to buy the image |
| 15:11.53 | sambal | cq. license |
| 15:13.06 | riksta | sambal: i wonder how many people actually do ;P |
| 15:13.13 | sambal | sssssst |
| 15:13.51 | ariel_ | Jearil, Cisco has it available to download if you have a service contract with them. But you are also suppose to pay for the lisc to use sip. |
| 15:14.37 | Jearil | alright.. thanks |
| 15:14.45 | *** join/#asterisk X-Gen (~x-gen@rrba-146-124-131.telkomadsl.co.za) |
| 15:14.52 | Jearil | can anyone recommend a good office IP phone that doesn't force you to buy a license to actually use it? |
| 15:15.03 | eKo1 | These FXO gateways are such a fucking headache. Argh! |
| 15:15.22 | sambal | Jearil: snom |
| 15:15.27 | epoch | Jearil: polycom soundpoint ip500 |
| 15:15.32 | ariel_ | I think that most just get the service contract and then download the sip which is not in there view correct. But it's expensive to get 98 dollars for the lisc thne another 9 for the service contract. |
| 15:15.35 | ariel_ | each. |
| 15:15.47 | CosmicRay | Jearil: SPA-841 seems to be highly recommended around these parts |
| 15:15.47 | Jearil | hmm, the ip500 we got to test had no sip software on it |
| 15:16.05 | epoch | Jearil: you can download it off the net... get the url from the wiki |
| 15:16.06 | Jearil | CosmicRay: alright.. I'll look into that.. Thank you :) |
| 15:16.10 | ariel_ | Sipura, Snom, Polycom all are good in my view Polycom better then the others. |
| 15:16.48 | *** join/#asterisk zno (~chatzilla@ip-160-79-174-100.autorev.intellispace.net) |
| 15:16.56 | CosmicRay | Jearil: I just ordered my SPA-841 from here: http://www.telephonyware.com/telephonyware/tw00113.html |
| 15:17.00 | nestAr | i'm still pissed off at Polycom |
| 15:17.01 | CosmicRay | Jearil: $85 |
| 15:17.07 | nestAr | i want to reprogram my softkeys |
| 15:17.14 | zno | CosmicRay: I had a bad experience with those |
| 15:17.15 | nestAr | and they said no |
| 15:17.19 | CosmicRay | zno: oh? |
| 15:17.25 | epoch | nestAr: what do you want them to do? |
| 15:17.36 | zno | CosmicRay: Half of my order of 841s stopped working |
| 15:17.42 | zno | probably a firmware issue |
| 15:17.42 | ariel_ | I have a 841 working great. |
| 15:17.50 | nestAr | when i'm on a call, i wanted to remove the soft button for hold and replace it with transfer |
| 15:17.56 | CosmicRay | zno: did you try upgrading to the latest firmware? |
| 15:18.05 | nestAr | since there's a big hold button at the bottom of the phone |
| 15:18.16 | zno | CosmicRay: Didn't bother, got SNOM 190s instead |
| 15:18.17 | Eight | The only point I have against the 841 is it doesn't have a 'pass through' switch =/ |
| 15:18.30 | nestAr | pass through is good |
| 15:18.49 | nestAr | i originally wanted two lans.. |
| 15:18.55 | nestAr | a phone lan and a pc lan |
| 15:18.58 | Eight | nestAr: ya. |
| 15:18.58 | nestAr | then i got lazy |
| 15:19.01 | Eight | nestAr: heh. |
| 15:19.08 | Eight | no no |
| 15:19.11 | Eight | you got "Cost effective" |
| 15:19.15 | nestAr | oh right |
| 15:19.17 | epoch | heh |
| 15:19.17 | nestAr | good one! |
| 15:19.35 | epoch | seriously, different LANs for phone & PC is a huge pain in the ass |
| 15:19.42 | CosmicRay | zno: $230 vs. $85 is a big leap |
| 15:19.50 | `Sauron | epoch: Not really |
| 15:19.51 | epoch | how many offices do yo know where they have multiple LAN jacks in each office? |
| 15:19.55 | `Sauron | Just depends on your edge gear |
| 15:20.05 | `Sauron | and then you just run vlan trunks everywhere |
| 15:20.15 | Eight | hmm, y'know what would be handy... |
| 15:20.20 | nestAr | well, our office network was a nightmare |
| 15:20.22 | CosmicRay | zno: I'd have to have four spa-841s fail before that would break even |
| 15:20.23 | zno | CosmicRay: Losing time because half your phones don't work is even more expensive |
| 15:20.23 | X-Gen | do 90% of sip phones have mini-100Mbit hubs in them ? |
| 15:20.26 | Eight | I remember seeing a 2 or 4 port switch *in* a jack, a while ago. |
| 15:20.36 | Eight | X-Gen: quite a few, not 90% though. |
| 15:20.41 | nestAr | i've removed 7 hubs, and 2 switches that were just unneeded |
| 15:20.53 | epoch | Eight: yeah, those are pretty cool |
| 15:20.54 | CosmicRay | X-Gen: Be careful, many have 10Mbit hubs! |
| 15:21.03 | CosmicRay | X-Gen: some don't have them at all. read the specs. |
| 15:21.15 | X-Gen | urg 10Mbit, i thought that died with rs-232 like 10yrs ago |
| 15:21.26 | CosmicRay | so did I |
| 15:21.28 | Eight | http://www.3com.com/products/en_US/detail.jsp?tab=features&pathtype=purchase&sku=WEBBNCNJSYS |
| 15:21.28 | nestAr | yeah.. the Budgetone is 10mbit hub |
| 15:21.40 | nestAr | it's got a realtek 8019 chip in it, i believe |
| 15:21.45 | X-Gen | so it the phone done have a mini-hub in it then u need a dedicated port on the hub |
| 15:21.58 | eKo1 | The spa-1001 is 10mbit |
| 15:22.14 | *** join/#asterisk MikeJ_ (~icechat5@65.170.43.34) |
| 15:22.31 | nestAr | the Polycom phones all have 100mbit switched passthru |
| 15:22.34 | nestAr | and they work well |
| 15:22.52 | epoch | they do VLAN too, I think |
| 15:22.56 | zno | For those who use debian + asterisk: Is there any way of getting zaptel to compile without building a custom kernel? |
| 15:22.56 | nestAr | i play CounterStrike through one all the time. ;D |
| 15:22.58 | eKo1 | I think 10mbit is enough for a phone. |
| 15:23.13 | Eight | eKo1: but not necessarily for the computer you want to connect to the other side of it. |
| 15:23.22 | ariel_ | eKo1, yes but sometimes you need more for the 2nd device connected to the phone. |
| 15:23.31 | nestAr | eKo1: phone, yes... but when you're pluging the PC Into the phone and the phone into the network.. |
| 15:23.31 | eKo1 | What 2nd device? |
| 15:23.34 | nestAr | no. |
| 15:23.46 | ariel_ | Networks --- Phone -- PC |
| 15:24.01 | eKo1 | I though it was Net -- PC -- phone. |
| 15:24.10 | epoch | uh, no |
| 15:24.11 | zno | the phone acting as a middleman between the net and pc is unacceptable. You lose your network when you reboot the phone. |
| 15:24.19 | bjohnson | eKo1: you're talking about something different |
| 15:24.19 | ariel_ | most pc's don't have a switch or hub on it. |
| 15:24.20 | X-Gen | how does 100mbit switched passthrough work ? when the phone wants to transmit, it just tells the other device 'there are collisions on the network...wait till i'm done' |
| 15:24.25 | eKo1 | like the spa-2100? |
| 15:24.29 | epoch | zno: er, for a split second, maybe... |
| 15:24.32 | bjohnson | eKo1: they are talking about phones with a second nic |
| 15:24.36 | bjohnson | yes like the 2100 |
| 15:24.39 | epoch | zno: I've never had problems when rebooting my polycom |
| 15:24.45 | ariel_ | eKo1, spa 2100 has a router not a swtich |
| 15:24.50 | epoch | also in production, how often do you really need to reboot your phone? |
| 15:24.59 | sambal | true :) |
| 15:25.04 | bjohnson | it has a second cat5 port for a computer |
| 15:25.04 | zno | well during a configuration change |
| 15:25.04 | eKo1 | epoch: every week at least. |
| 15:25.14 | nestAr | polycom switch survives a reboot |
| 15:25.18 | epoch | eKo1: your production environment sucks ;) |
| 15:25.28 | nestAr | only a power failure breaks it |
| 15:25.37 | ariel_ | then that is good of polycom because the cisco don't |
| 15:25.37 | epoch | and that's where PoE comes in :) |
| 15:25.43 | eKo1 | I have power failures up the wazoo |
| 15:25.43 | zno | half the apps on my pc have to be restarted when i reboot my phone |
| 15:25.51 | epoch | eKo1: get a UPS ;) |
| 15:26.05 | eKo1 | epoch: I have one but it's not a true online UPS. |
| 15:26.15 | epoch | well I see how that's unacceptable then... |
| 15:26.28 | eKo1 | I needs me PoE dang it. |
| 15:26.33 | epoch | I've got a polycom backed up by UPS in our server room ;) |
| 15:26.37 | nestAr | i need PoE too |
| 15:26.39 | epoch | with pseudo-PoE, heh |
| 15:26.44 | nestAr | lol |
| 15:26.50 | eKo1 | Only polycom and cisco do PoE right? |
| 15:27.00 | nestAr | PoE wasn't cost effective for us at this point |
| 15:27.01 | epoch | eKo1: I'm sure other vendors do too |
| 15:27.06 | Eight | Those 3Com 4-port PoE powered in-jack-switches are about 50-60/ea |
| 15:27.32 | eKo1 | I wish my SPAs were PoE. |
| 15:27.33 | zno | I have yet to find a good linux distro that works well with asterisk. My adventures in debian are petering out. I'm sick of building a custom kernel because zaptel relies on a USB module that requires custom buildling. |
| 15:27.54 | epoch | nestAr: you should see our server room ;) there's 3 power bars with a whole whack of polycom power supplies, plugged into the inline power injection cables ;) |
| 15:27.56 | Eight | zno: switch to kernel 2.6, ZTdummmy doesn't require the USB timing. |
| 15:28.06 | epoch | PoE, with no extra expense! ;) |
| 15:28.13 | zno | Eight: doesn't 2.6 have problems as well? |
| 15:28.26 | nestAr | epoch: lolol |
| 15:28.34 | Eight | zno: perhaps, but I haven't had trouble myself. |
| 15:28.45 | nestAr | epoch: you using all ip500's? |
| 15:28.48 | epoch | nestAr: yep |
| 15:28.59 | zno | Eight: so basically any distro that uses 2.4 will have this USB dependency |
| 15:29.01 | nestAr | most of our deployment is ip300 |
| 15:29.03 | nestAr | with some 500 |
| 15:29.12 | epoch | the 300's basically the same thing, right? |
| 15:29.13 | nestAr | we're "ballin' on a budget" |
| 15:29.15 | epoch | except for the mic? |
| 15:29.19 | Eight | zno: so long as you want meetme and don't have any actual zaptel hardware. That's my understanding. |
| 15:29.20 | nestAr | well, they don't use the power cable |
| 15:29.25 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 15:29.25 | epoch | and 2 lines as opposed to 3? |
| 15:29.29 | epoch | oh? |
| 15:29.32 | nestAr | you plug the ac adapter into the phone |
| 15:29.34 | nestAr | not the cbale |
| 15:29.35 | nestAr | cable |
| 15:29.38 | epoch | wtf |
| 15:29.40 | epoch | that's weak |
| 15:29.41 | nestAr | sucks for sure |
| 15:29.45 | epoch | so the 300 can't handle PoE at all? |
| 15:30.12 | nestAr | not without some wiring |
| 15:30.12 | nestAr | :x |
| 15:30.12 | epoch | I think we just recommended 40 ip300s to a potential customer |
| 15:30.32 | zno | Eight: that's good info. I'm gonna have to try 2.6 just to ease the pain. |
| 15:30.36 | johnnyb | nestAr: The specs say it can do either. |
| 15:30.37 | nestAr | i've done some ghetto PoE for some axis cams |
| 15:30.49 | nestAr | johnnyb: maybe i should try it.. |
| 15:31.04 | nestAr | in any event..those cables are like $30 a pop and the IP300s don't include them |
| 15:31.17 | johnnyb | What is a good PoE switch? |
| 15:31.21 | Eight | zno: fwiw, I'm using Fedora. I'm not really a linux person (FreeBSD, OSX, primarily), but this seems to be working out for me. |
| 15:31.41 | johnnyb | nestAr: what cables? I thought PoE just went through the normal ethernet cables. |
| 15:31.43 | zno | Eight: me too, I'm a FreeBSD guy, but had to use linux for zaptel/meetme |
| 15:31.57 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 15:32.02 | Eight | zno: seems everyone in here that's bitching about Linux is a BSD person =) |
| 15:32.05 | nestAr | johnnyb: if you're using a POE switch |
| 15:32.10 | nestAr | i don't have one of those.. |
| 15:32.22 | johnnyb | nestAr: I see. |
| 15:32.35 | Eight | 139 for the IP300 w/10|100 switch is neat. |
| 15:32.37 | johnnyb | nestAr: one of the reasons I went with budgetone is that I don't have one either. |
| 15:32.52 | zno | Eight: seems nothing ever gets close to freebsd's package management and kernel source syncing |
| 15:33.00 | TheBear | Anyone using the festival-weather-script.pl ? (1) I don't have the tts directory (2) I'm getting an error "Mar 11 10:31:26 WARNING[29819]: file.c:486 ast_openstream_full: File tts/tts-d463cc1961b5a8b0e78842ce76252428 does not exist in any format" any advice ? |
| 15:33.05 | Eight | zno: ::sigh:: ya. |
| 15:33.25 | *** join/#asterisk X-Gen (~x-gen@rrba-146-124-131.telkomadsl.co.za) |
| 15:33.34 | X-Gen | thats a 1st, opera crashed |
| 15:35.25 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:35.35 | nestAr | johnnyb: here's the polycom PoE cable.. this is for use when you don't have a PoE switch.. |
| 15:35.38 | nestAr | http://gallery.wewt.net/albums/voip/P1000607.sized.jpg |
| 15:35.54 | nestAr | epoch: here's the back of the IP300 |
| 15:36.03 | nestAr | http://gallery.wewt.net/albums/voip/P1000606.sized.jpg |
| 15:37.27 | alexns | i noticed 2 specs on poe, one was cisco one was open spec |
| 15:38.11 | nestAr | epoch: i took an IP500 cable and plugged it into the IP300, it works |
| 15:38.17 | alexns | will cisco phones work with poe on adtran switch for example |
| 15:38.26 | TheBear | what is the diff. between Asterisk@home and asterisk ? |
| 15:38.55 | johnnyb | That's pretty cool. How much are such cables? |
| 15:38.55 | Nugget | one is a livecd linux distro with asterisk as its primary feature. the other is asterisk. |
| 15:39.03 | ariel_ | alexns, no cisco is the one that is not standard. |
| 15:39.13 | *** part/#asterisk Dirtysole (~Frank@cpe.atm0-0-0-1201091.0x503fc556.bynxx2.customer.tele.dk) |
| 15:39.35 | ariel_ | TheBear, asterisk@home is asterisk but with an ISO complete build of CentOS with Asterisk and amp and some other nice applications. |
| 15:39.44 | *** join/#asterisk sariabod (~sariabod@ip21.farheap.net) |
| 15:40.01 | TheBear | ariel_: what are the apps, would they cause agi scripts to fail ? |
| 15:40.18 | bjohnson | does anyone have a search and replace script for linux that will take a filename pattern? |
| 15:40.35 | Nugget | you mean like sed? |
| 15:40.40 | Nugget | what do you want to do? |
| 15:40.42 | ariel_ | Well if yoru script is not working it's most likly to do with user rights. It changes allot of things in that way. |
| 15:40.46 | mountie | bjohnson: like find ... -exec sed ... |
| 15:40.47 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
| 15:40.47 | bjohnson | yes like sed |
| 15:40.48 | Nugget | or do you mean renaming files? |
| 15:41.02 | bjohnson | but a bash script or something to make it easier |
| 15:41.12 | Nugget | what's not easy about sed? :) |
| 15:41.23 | bjohnson | like sed_replace texttofind texttoinsert filenamematch |
| 15:41.32 | Nugget | you don't need a script to do that |
| 15:41.39 | *** join/#asterisk sariabod (~sariabod@ip21.farheap.net) |
| 15:41.42 | Nugget | use mountie's suggestion. find and sed in conjunction. |
| 15:41.54 | bjohnson | it's sed that is the problem |
| 15:41.57 | Nugget | why? |
| 15:42.04 | bjohnson | too complicated |
| 15:42.28 | Nugget | but all your theoretical script can do is search/replace. that's not complicated. |
| 15:42.59 | bjohnson | example? |
| 15:43.07 | Nugget | if you only want one feature of sed, it's silly to get hung up on all the features you don't plan to use. |
| 15:43.20 | Nugget | find . -name foo\* -exec sed 's/texttofind/texttoinsert/g' |
| 15:43.34 | *** part/#asterisk dubphil (~dubphil@80.124.137.201) |
| 15:43.36 | bjohnson | thnx |
| 15:44.02 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 15:44.18 | Nugget | it would have been easier if you'd just asked how to use sed instead of making us play 20 questions. |
| 15:44.25 | PBXtech | is RXfax buggy? im getting a lot of half page fax's |
| 15:44.29 | nestAr | lol |
| 15:44.52 | bjohnson | Nugget: sed is just one solution .. not the only one |
| 15:44.56 | ariel_ | PBXtech, the reply is yes |
| 15:45.07 | Zeeek | seconded |
| 15:45.12 | PBXtech | is there something more solid? |
| 15:45.21 | Zeeek | a $35 fax machine |
| 15:45.27 | Nugget | sure, but you made it sound like sed was not the right solution when in reality you were just scared of sed because you don't know how to google. :) |
| 15:45.30 | PBXtech | heh umm |
| 15:46.32 | pcm | fax machines are $50 |
| 15:46.39 | ariel_ | PBXtech, I have a zap port on my system I detect fax then send it to my fax machines. Works great. |
| 15:46.51 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 15:47.31 | Zeeek | best solution bar none |
| 15:47.48 | PBXtech | for a single solution yea |
| 15:47.57 | ariel_ | even if it has no papers it can store up to 60 pages. |
| 15:47.57 | *** join/#asterisk mikegrb (~michael@thegrebs.com) |
| 15:48.01 | Zeeek | is to make fax obsolete |
| 15:48.30 | ariel_ | I do more in the way of printing to a pdf and then sending them via email. Works great. |
| 15:48.49 | puppet | How many here are intrested in a PHP API against the Mananger? ;p |
| 15:49.15 | alexns | i hate fax |
| 15:49.27 | alexns | but you need it for signatures.. |
| 15:49.30 | *** join/#asterisk BurnedOutGeek (~BurnedOut@216.215.202.4.nw.nuvox.net) |
| 15:49.36 | alexns | sure you could scan & email |
| 15:49.53 | puppet | scan & email > fax |
| 15:50.24 | fotang | ariel_: where do i put `nat=no' as you suggested? |
| 15:51.07 | alexns | they need to make a "fax" that scans & emails, something better than hp |
| 15:51.14 | alexns | need a simpler commercial version |
| 15:51.51 | johnnyb | Xerox already has one, I think. It scans and send PDFs. |
| 15:51.53 | *** join/#asterisk nwhit (~chatzilla@67.104.119.206.ptr.us.xo.net) |
| 15:52.05 | johnnyb | My dad uses it to send me stuff all the time. |
| 15:52.14 | alexns | thats decent |
| 15:52.54 | alexns | i just think its funny that the biggest problem to voip is a stupid analog fax machine |
| 15:53.02 | alexns | ironic in a way |
| 15:53.25 | alexns | whats the status of asterisk t.38 |
| 15:53.31 | *** join/#asterisk coppice (~chatzilla@111.196.17.210.dyn.pacific.net.hk) |
| 15:53.58 | nwhit | hello all... can anyone give me some help... my asterisk server has a public IP and is not behind NAT, I have a few sip devices that are behind NAT that after some time asterisk can't find them to route a call until I make an outbound call, any suggestions? |
| 15:55.46 | alexns | sure |
| 15:56.15 | alexns | do you have nat=yes under the extensions in sip |
| 15:56.28 | nwhit | yes |
| 15:56.29 | *** join/#asterisk channan (~channan9@66.180.121.185) |
| 15:56.45 | Zeeek | sounds more like the registration interval is too long |
| 15:56.48 | *** join/#asterisk Pazak (Kino@user-0cev2l9.cable.mindspring.com) |
| 15:56.55 | Zeeek | make the phones register more often |
| 15:57.02 | alexns | what kind of phones |
| 15:57.08 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 15:57.08 | nwhit | zeek... suggested value? |
| 15:57.17 | nwhit | sipura and snom |
| 15:57.19 | Eight | is there a way to tell asterisk how long to 'hold' registrations? |
| 15:57.29 | Zeeek | nwhit are you forwarding ports on the clients side? |
| 15:57.45 | nwhit | no |
| 15:57.59 | Zeeek | try 1 minute - if that helps, try 2 etc |
| 15:58.16 | bjohnson | how can I feed a backslash to sed? ie find . -name foo\* -exec sed 's/texttofind/newsubdir\/texttoinsert/g' ? |
| 15:58.33 | nwhit | what is weird, is that I have nat=yes set in sip.conf, but when I do a sip show peers it shows nat = N |
| 15:58.42 | *** join/#asterisk file (~file@251.134.218.209.transedge.com) |
| 15:58.47 | nwhit | zeeek: ok |
| 16:00.08 | Zeeek | someone just told me there is an issue of bad quality when talking on IAX to ZAP. Can someone confirm this? |
| 16:00.25 | alexns | in polycom phones you change reg time in xml config file |
| 16:00.34 | Corydon-w | Zeeek: nope, never had that problem |
| 16:00.49 | Zeeek | Corydon-w what IAX equip? |
| 16:00.53 | alexns | does snom phone use config files on tftp/ftp ? |
| 16:01.04 | Corydon-w | Zeeek: Asterisk, of course |
| 16:01.19 | Corydon-w | From one Asterisk system to another |
| 16:01.29 | Zeeek | I meant an IAXy or IAX phone talking to * to a ZAP phone |
| 16:01.42 | Zeeek | I have two cases of thi. |
| 16:01.45 | Corydon-w | Nope, works well enough for fax |
| 16:02.13 | Corydon-w | That right there should tell you there's no problem with audio, given how sensitive fax machines are |
| 16:02.27 | nwhit | alexns: not sure... but they have a good web interface |
| 16:02.31 | Zeeek | I just was talking an IAXy to my wife at the office. I heard her perfectly but she was having trouble with distortion and noises. I'm about 35 ms away |
| 16:02.33 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 16:02.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:03.15 | Zeeek | And I experience breaking up between IAX phone and ZAP phone on same asterisk box, same side of same network, <2ms |
| 16:03.15 | Corydon-w | How consistent was the 35ms? Is that also true for packets larger than ICMP echo? |
| 16:03.54 | Corydon-w | Might be your box is overloaded... or your network... there's too many variables to blame it on iax |
| 16:04.01 | Zeeek | the 35 never goes above 50 - granted if it is just one time, it's meaningless. Also, is she just trying to diss my asterisk? |
| 16:04.23 | Zeeek | box has only two channels talking at any one time |
| 16:04.34 | Zeeek | one ZAP, one IAX |
| 16:04.43 | Corydon-w | What's the speed, and what else is running on that box? |
| 16:04.44 | Zeeek | or SIP-SIP, SIP-IAX etc |
| 16:04.52 | Corydon-w | Also, is the Zap analog or digital? |
| 16:05.23 | Zeeek | nothing at all is running on the box, not even Apache, Pentium III-800, 512 Meg RAL, TDM400P |
| 16:05.34 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 16:05.39 | Corydon-w | FXO or FXS interface? |
| 16:06.18 | Zeeek | 3xFXS oon the TDM, also 2x X100P but no calls at the time of problem |
| 16:06.44 | bjohnson | all the sed experts have gone quiet |
| 16:06.54 | puzzled | anyone using zaptel with udev mind msg'ing me the output of ls -l /dev/zap |
| 16:06.55 | Zeeek | Corydon-w anyway, I never heard of that before so I was curious |
| 16:06.56 | Corydon-w | You know you're exceeding the number of Digium cards recommended in a single system, right? |
| 16:07.18 | Zeeek | no I never saw anywhere that three cards in a system with nothing else was too many |
| 16:07.34 | puzzled | 2 is the magic number afaik |
| 16:07.56 | Corydon-w | I've always heard 2, and it's due to the number of interrupts generated per card, not total system load |
| 16:08.05 | Zeeek | why would this make trouble only re IAX and not the other |
| 16:08.11 | *** join/#asterisk n3tar (~geno@201.254.30.251) |
| 16:08.14 | n3tar | hi |
| 16:08.26 | Corydon-w | It shouldn't |
| 16:08.33 | Zeeek | when a card is waiting does it have less interrupts that when it's busy? |
| 16:08.41 | Corydon-w | What codec are you using? |
| 16:08.46 | Zeeek | IOW, do IRQ increase with traffic? |
| 16:08.56 | Corydon-w | Nope, same IRQ load no matter what |
| 16:09.00 | *** join/#asterisk folsson (~filip@h87n2fls31o985.telia.com) |
| 16:09.06 | Zeeek | not a huge choice with IAXy, I think ulaw |
| 16:09.20 | Corydon-w | The only thing that changes with load is the number of bytes read per interrupt |
| 16:09.36 | Zeeek | seems logical enough |
| 16:10.21 | Zeeek | in fact it isn't IAX, but IAX<->ZAP that was mentioned |
| 16:10.33 | Corydon-w | Well, maybe ulaw is taking up enough bandwidth remotely to make a difference. You might try g726 for a comparison |
| 16:10.34 | Zeeek | Like I say, I never heard anyone complain about that |
| 16:11.02 | Zeeek | But the same IAXy talking to SIP on the same connection has no problems |
| 16:11.16 | nwhit | thanks for your help... :) I think that is going to take care of it |
| 16:11.34 | Zeeek | nwhit now you can try increasing the register time if you need to |
| 16:12.01 | nwhit | ok |
| 16:12.38 | Darwin35 | Darwin35 ok 1.0.6 i get no audio but i get dial tone |
| 16:12.38 | Darwin35 | Darwin35 and this is with dialing internal extensions |
| 16:12.38 | Darwin35 | Darwin35 grr |
| 16:12.57 | Zeeek | dial tone from where? |
| 16:13.08 | Zeeek | ZAP? |
| 16:13.12 | Darwin35 | if i dial from my sip to my test extesions |
| 16:13.21 | Darwin35 | i get no audio |
| 16:13.22 | Zeeek | sip phone does the dialtone |
| 16:13.58 | Darwin35 | i wtch the call it shows it goes threw no errors |
| 16:14.17 | Darwin35 | 1001/1001 192.168.1.2 D N 255.255.255.255 5060 Unmonitored |
| 16:14.17 | Darwin35 | <PROTECTED> |
| 16:14.17 | Darwin35 | <PROTECTED> |
| 16:14.17 | Darwin35 | <PROTECTED> |
| 16:14.17 | Darwin35 | <PROTECTED> |
| 16:14.17 | Zeeek | so you won the war but everyone was killed |
| 16:14.18 | Darwin35 | <PROTECTED> |
| 16:14.20 | Darwin35 | <PROTECTED> |
| 16:14.22 | Darwin35 | <PROTECTED> |
| 16:14.24 | Darwin35 | <PROTECTED> |
| 16:14.26 | Darwin35 | <PROTECTED> |
| 16:14.28 | Darwin35 | <PROTECTED> |
| 16:14.30 | Darwin35 | <PROTECTED> |
| 16:14.34 | Darwin35 | <PROTECTED> |
| 16:14.36 | Darwin35 | sorry |
| 16:14.38 | Darwin35 | sorry was pasting to pastebin |
| 16:14.47 | Darwin35 | but hit wrong window |
| 16:14.56 | Zeeek | well no need now :) |
| 16:15.02 | puzzled | lol |
| 16:15.07 | Eight | heh |
| 16:15.09 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 16:15.12 | Darwin35 | sorry |
| 16:15.21 | Zeeek | so where is the sip phone on the network? |
| 16:15.23 | Eight | Darwin35: happens to the best (and it would seem worst) of us =) |
| 16:15.23 | Darwin35 | but i get no audio |
| 16:15.36 | Zeeek | I did that yesterday |
| 16:15.39 | Darwin35 | internal network |
| 16:15.45 | file | Darwin35: try 1.0.7 |
| 16:16.08 | Zeeek | Darwin also is the Grandstream set to start RTP at 10000 ? |
| 16:16.11 | Zeeek | mine are |
| 16:16.25 | Darwin35 | ok |
| 16:16.35 | Zeeek | local RTP port: |
| 16:16.40 | Zeeek | just under SIP port |
| 16:16.53 | *** join/#asterisk Derkommissar (~alberto@65.165.85.132) |
| 16:18.58 | *** join/#asterisk p0lar (~p0lar@64.254.225.62) |
| 16:19.00 | bjohnson | turns out the command given by Nugget is wrong. Should be find . -name foo\* -exec sed 's/texttofind/texttoinsert/g' {} \; |
| 16:19.06 | bjohnson | :q! |
| 16:19.40 | file | there once was a remote around here... but where oh where did it go |
| 16:20.38 | Derkommissar | For some reason h.323 calls get rejected ven before hitting the dialplan :-/ |
| 16:21.02 | Beirdo | file: check under yer butt? |
| 16:21.15 | file | I never had it |
| 16:21.20 | file | I think twisted has it |
| 16:21.28 | Beirdo | ahh |
| 16:21.51 | file | his cellphone has been going off a bit too |
| 16:21.56 | marlowe | Does anyone know if sImpletelecom offers inbound caller id info? |
| 16:22.11 | Darwin35 | ok |
| 16:22.27 | Zeeek | ya Darwin? |
| 16:22.30 | Darwin35 | well I have to check the phone from another system |
| 16:22.33 | Derkommissar | 0:42.690 H225 Answer:9f18fe0 transports.cxx(1150) H225 Signal channel stopped on first PDU. |
| 16:22.41 | Derkommissar | why does that happen |
| 16:24.20 | phantasis | how do I get IAX2 soft phones to work with AMP |
| 16:25.17 | X-Gen | hey hey hey, its friday ! |
| 16:26.52 | nestAr | hey hey what can i do.. i gotta woman.... she won't stay true.. |
| 16:27.10 | phantasis | kill her :) |
| 16:27.25 | phantasis | anyone have any experience with AMP? |
| 16:27.44 | X-Gen | nestAr: time 2 upgrade |
| 16:29.21 | nestAr | ;) |
| 16:31.38 | CosmicRay | assuming you meant asterisk management portal |
| 16:34.30 | xstaci | what is a DID route? and what would one look like for say vonage as a sip trunk |
| 16:34.47 | CosmicRay | DID means a dial-in |
| 16:34.56 | CosmicRay | so that gives you a "real" pstn number |
| 16:35.17 | CosmicRay | usually then it would be routed over the internet to your installation |
| 16:35.39 | CosmicRay | look here: http://www.voip-info.org/wiki-VOIP+Service+Providers+Residential |
| 16:36.30 | *** join/#asterisk aricon (equitable@chaosiv.buh.org) |
| 16:37.12 | aricon | hi |
| 16:37.13 | xstaci | ahh, so vonage is a no go? |
| 16:37.16 | TheBear | in festival.conf the "tts_textasterisk" is that a file that should be on the system ? |
| 16:37.17 | *** join/#asterisk greendisease (~jack@greendisease.fedora) |
| 16:38.02 | aricon | I see on the website that dialogic hardware based upon the JCT bus is supported, but will any of the older SC bus devices work? |
| 16:39.52 | *** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 16:40.24 | aricon | and if so what version of the drivers is required.. 5.1, 5.1SR1, 5.1SR1 with FP1sp3? |
| 16:41.20 | CosmicRay | xstaci: I don't know, but I'd consider it a bad sign that they supply a "locked" nuit |
| 16:41.28 | eKo1 | Hmm...I think my dslam just kicked the bucket. |
| 16:41.30 | CosmicRay | xstaci: there are plenty of cheaper & better competitors anyway |
| 16:41.39 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 16:41.54 | Goshen | Hey, does Broadvoice pass the Caller ID you set? |
| 16:41.59 | coppice | aricon: JCT isn't a bus. its just a range of cards. the older ones are not capable of doing VoIP, and even the JCT ones kind of suck |
| 16:42.02 | *** join/#asterisk ikey (~kirankuma@220.226.16.166) |
| 16:42.20 | ikey | hi can any one help in setting up sip and h323 in asterisk |
| 16:42.26 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 16:42.54 | CosmicRay | I have Asterisk on my local network, and a sjphone pcphone. When I dial the asterisk server's echo test feature, the echo back is at least half a second delayed. I'm using ulaw, and can't figure out what could be causing that delay. |
| 16:42.55 | CosmicRay | any ideas? |
| 16:43.31 | Goshen | CosmicRay: on your lan? |
| 16:43.34 | CosmicRay | yes |
| 16:43.41 | Goshen | hmm |
| 16:43.41 | CosmicRay | just a 100Mb switch between them. |
| 16:43.49 | Goshen | have you tried FWD echo test? |
| 16:43.50 | aricon | ok so no ISA SC bus cards will work? |
| 16:43.57 | aricon | or just not with VOIP |
| 16:44.03 | CosmicRay | Goshen: yes, it is (understandably) worse |
| 16:44.15 | CosmicRay | though I was still connecting via sterisk |
| 16:44.34 | Goshen | Have you tried an IAX client? |
| 16:44.36 | Goshen | ~firefly |
| 16:44.37 | jbot | i heard firefly is http://virbiage.com/firefly/download/firefly-thirdparty.exe |
| 16:44.59 | julianjm | nwhit: use qualify=15000 so that each 15 seconds, asterisk "pings" your clients an keep the firewall open |
| 16:45.45 | CosmicRay | Goshen: no, I haven't. I'm using Linux, so there are not a lot of good choices. otoh, I will be using SIP hardphones anyway, so I'd really want to make SIP work |
| 16:45.56 | CosmicRay | Goshen: out of curiosity, does your nick refer to the town in northern indiana? |
| 16:46.09 | Goshen | nope |
| 16:46.23 | Goshen | There is a town of Goshen in many states |
| 16:46.32 | Goshen | It just came to me one day |
| 16:46.41 | Goshen | I didn't even know about the Bible reference or the towns |
| 16:47.24 | xstaci | cosmicray: what voip provider do you suggest for connecting to an * box? |
| 16:47.51 | *** join/#asterisk jjg (~jjg@63.166.156.128) |
| 16:47.53 | jjg | hi |
| 16:48.11 | CosmicRay | xstaci: I don't really know yet. I've posted some suggestions I've gleaned from the wiki and the community on my blog here: http://changelog.complete.org/node/252 |
| 16:48.22 | CosmicRay | xstaci: I have some serious concerns about voipjet's terms of service, though |
| 16:48.26 | jjg | anyone know if a single xeon 2.8 GHz with Gig of RAM is enough to run 16 simultaneous outbound voip calls via sip? |
| 16:48.36 | eKo1 | Somebody needs to change the channel topic already. |
| 16:49.03 | eKo1 | jjg: Sure. |
| 16:49.54 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
| 16:50.25 | ikey | jjg: it can take up 120 calls simultaniously |
| 16:50.35 | *** join/#asterisk jmacz (~jmacz@201.245.167.104) |
| 16:50.55 | *** join/#asterisk NK123 (~p645@24.163.78.12) |
| 16:51.11 | jjg | ikey : 120 calls on a single proc !? |
| 16:51.20 | jjg | that sounds a bit off |
| 16:51.21 | NK123 | ikey |
| 16:51.35 | jjg | i would think maybe around 30 calls per processor |
| 16:51.47 | jjg | anyone actually running 120 cps on a single proc? |
| 16:52.11 | ikey | jjg: yeah it can take we do lot of dialouts on PSTN it self |
| 16:52.34 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 16:52.44 | jjg | but dialout on the pstn, isn't the dsp being offloaded to a external card? |
| 16:52.48 | ikey | we do ... |
| 16:52.54 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:53.08 | jjg | ikey : what is your HW config...ie, what brand and speed of proc |
| 16:53.35 | file | jjg: what codecs are you doing? |
| 16:53.42 | ikey | yep Still we use only 35-40 percent of CPU |
| 16:53.45 | eKo1 | I don't know of anybody pushing 120 concurrent calls. |
| 16:53.58 | AgiNamu | Oh yea! IAX2 hardphone with iLBC support! booya! |
| 16:54.08 | ikey | jjg: on SIP it should be comfortable with 120 |
| 16:54.16 | eKo1 | I have at most 20 active channels and * consumes less that 20% of the CPU. |
| 16:54.34 | jjg | file : i would probably use gsm |
| 16:54.35 | AgiNamu | jjg: What kind of transcoding? |
| 16:54.40 | Juggie | 20 active chans shoudnt use any cpu |
| 16:54.44 | Juggie | unless u are using transcoding |
| 16:54.45 | AgiNamu | if you're doing pure voip passthru, 20 channels is nothing |
| 16:54.47 | Juggie | which is not reccomended |
| 16:54.57 | Juggie | avoid it if you can |
| 16:55.55 | eKo1 | I try to. |
| 16:56.37 | eKo1 | I have 10 active channels and my CPU load is at 3%. |
| 16:56.40 | *** join/#asterisk Tr0j4N (b3nz3r@pool-151-204-65-164.delv.east.verizon.net) |
| 16:56.40 | jjg | eko1 : what kind of machine are you using...ie cpu type/speed and machine brand |
| 16:57.05 | eKo1 | 1 GB RAM |
| 16:57.10 | eKo1 | SMP Pentium 4. |
| 16:57.21 | AgiNamu | jjg: what are you doing? |
| 16:57.23 | Sedorox | smp as in HT? |
| 16:57.31 | jjg | eKo1 : SMP, so dual procs? |
| 16:57.47 | eKo1 | 2 CPUs at 2.8 GHz. |
| 16:57.51 | jjg | AgiNamu : i want up to 24 concurrent outbound |
| 16:58.02 | jjg | eKo1 : i was asking about single procs |
| 16:58.25 | jjg | AgiNamu : gsm, prolly IAX2 trunking via nufone if they support it |
| 16:58.30 | eKo1 | All my * servers are 2 cpu machines. |
| 16:58.50 | jjg | eKo1 : ok, but i was asking bout single. but thanks for the input |
| 17:00.26 | *** join/#asterisk Weezey (Weezey@lan6.LO.iasl.com) |
| 17:00.29 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-221-69.dsl.scarlet.be) |
| 17:00.34 | *** join/#asterisk Nohair (~Jez@srscomp.demon.co.uk) |
| 17:00.38 | mrgoby | i'm having some issues with the manager interface. it is transfering my call to the context i want, and the extension i want, but it is going the priority i specify minus one, and it isnt actually executing that command, just hanging, even though "show channels" from the CLI shows it is on that extension and priority |
| 17:00.43 | Weezey | how do I set the incoming caller ID for an h323 connection? |
| 17:01.10 | Weezey | Right now I get the user's name, sent by his end, but "asterisk" as the number. |
| 17:01.26 | *** join/#asterisk NetOfSickCoder (~um@200.121.129.178) |
| 17:01.52 | phantasis | how does asterisk handle smp? |
| 17:02.05 | *** join/#asterisk grumpie (~vijay@dsl093-139-122.sfo4.dsl.speakeasy.net) |
| 17:02.16 | Nohair | Hi I have a problem with transmit_register see http://pastebin.ca/7253 |
| 17:02.30 | CosmicRay | well, it appears to be multithreaded, so I would say: transparently. |
| 17:02.52 | *** part/#asterisk d1zzy (~dizzy@developer.user) |
| 17:02.56 | TheBear | in festival.conf the "tts_textasterisk" is that a file that should be on the system I can't find it on mine ? |
| 17:03.29 | ikey | yeah But frankly on quad e1 card of digium dialouts are pretty stable SIP we need to Try out the load |
| 17:03.29 | ikey | Eko1: how are u doing the load test on sip .... |
| 17:03.29 | ikey | and Can any one guide me to perfectly configure SIP channels for interoffice calling and Dailouts |
| 17:03.50 | *** join/#asterisk mesi (~player@dsl-082-083-152-197.arcor-ip.net) |
| 17:03.57 | mesi | re |
| 17:05.09 | johnnyb | ikey: what are you wanting? I just finished configuring my office w/ Asterisk |
| 17:05.35 | Nohair | any 1 help chan_sip.c:4187 |
| 17:05.40 | *** join/#asterisk shazb0t (~shazb0t@adsl-68-93-250-74.dsl.rcsntx.swbell.net) |
| 17:05.43 | johnnyb | I made heavy use of the defaults in asterisk - the stdexten macro is pretty good, so are their dialing contexts |
| 17:05.46 | phantasis | so Asterisk should work well on an 8 UltraSparc machine with 8GB of RAM? |
| 17:06.27 | ikey | Johnnyb: yeah we tried configuring the sip.conf |
| 17:06.34 | johnnyb | Nohair: you are leaving out considerable information about the error. |
| 17:06.41 | puppet | i came up with a new thing ;D |
| 17:06.45 | puppet | thebear: use AGI instead |
| 17:06.52 | johnnyb | ikey: what context are you putting the SIP phones in? |
| 17:06.54 | puppet | write a chan_shype.c |
| 17:06.54 | puppet | ;p |
| 17:07.00 | puppet | skype* |
| 17:07.06 | Nohair | johnnyb see http://pastebin.ca/7253 |
| 17:07.14 | johnnyb | ikey: Also, what extensions are you using? |
| 17:07.32 | johnnyb | Nohair: I did -- you left out several lines, I believe. Plus it would be good to have some context about what is happening. |
| 17:07.50 | ikey | Johnnyb: i used grandstream ip phones |
| 17:07.56 | phantasis | anyone know of a distro that is easy to install on a sparc machine? |
| 17:08.05 | johnnyb | ikey: what context did you put them in in your sip.conf? |
| 17:08.18 | ikey | Johnnyb: and configured a small code in extensions |
| 17:08.19 | spackle | johnnyB: how do people like asterisk in your office? |
| 17:08.35 | Nohair | johnnyb i will past some more. This errors happen when i start my asterisk box |
| 17:08.44 | johnnyb | spackle: They love it so far. On the other hand, our old system was at least 10 years old and sucked horribly. |
| 17:09.13 | Katty | hmm |
| 17:09.17 | spackle | Did the old system have Voicemail? |
| 17:09.48 | johnnyb | Spackle: it used to, until one day it started garbling 75% of the messages. |
| 17:09.59 | jjg | mrgoby : are you still there? |
| 17:10.05 | TheBear | puppet: I've tried the AGI didn't work reports 0,1,2,3,4 not the weather? so I'm trying a new script |
| 17:10.08 | Nohair | johnnyb see pastebin.ca/7254 |
| 17:10.10 | ikey | Johnnyb:i could see the sip phones registered on CLI |
| 17:10.29 | spackle | JohnnyB, nice, what manufacturer was that? |
| 17:10.39 | aricon | quit |
| 17:11.02 | johnnyb | spackle: AT&T |
| 17:11.10 | johnnyb | Nohair: what phones are you using? |
| 17:11.30 | puppet | thebear: agi works great here |
| 17:11.39 | spackle | partner system or something earlier? |
| 17:11.59 | Nohair | johnnyb that is the external connection to my sip provider all works just get the errors - any idea on the music on hold |
| 17:13.26 | jjg | ikey : just to clarify, you can get 120 cps on a single CPU? |
| 17:13.56 | Darwin35 | I need to do the calling card and the wakeup |
| 17:14.02 | Darwin35 | next |
| 17:14.09 | Darwin35 | I have to find them |
| 17:14.34 | Darwin35 | you must have zap for moh |
| 17:14.41 | puppet | darwin35: no u dont |
| 17:14.43 | Darwin35 | mpg123 is crap |
| 17:14.47 | Darwin35 | ? |
| 17:14.50 | puppet | darwin35: you need zap for meetme |
| 17:14.51 | ikey | <PROTECTED> |
| 17:14.51 | Darwin35 | since when |
| 17:14.52 | puppet | not for moh |
| 17:14.58 | Darwin35 | ahh ok |
| 17:15.02 | puppet | im running moh nice without zap |
| 17:15.08 | Darwin35 | I thought you had to have it for moh also |
| 17:15.11 | Darwin35 | ok |
| 17:15.22 | ikey | we do close about 8-9 Million dialouts per month |
| 17:15.45 | Darwin35 | madplay on fbsd is much better the mpg123 |
| 17:15.52 | ikey | jjg: we shifted from dialogic to this and this seems to be quite comfortable |
| 17:15.56 | Darwin35 | but I still have no audio |
| 17:16.01 | Darwin35 | grr |
| 17:16.44 | puppet | darwin35: you have to run mpg123 |
| 17:16.54 | *** part/#asterisk shazb0t (~shazb0t@adsl-68-93-250-74.dsl.rcsntx.swbell.net) |
| 17:17.04 | Darwin35 | no you dont |
| 17:17.11 | Darwin35 | madplay works |
| 17:17.16 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 17:17.35 | *** join/#asterisk shazb0t (~shazb0t@adsl-68-93-250-74.dsl.rcsntx.swbell.net) |
| 17:17.57 | Darwin35 | mpg123 othr then the one in base of * is screwed |
| 17:18.25 | puppet | doesnt madplay lagg, reset when itplay, stops playing etc? |
| 17:18.46 | Darwin35 | not had a issue yet |
| 17:19.01 | puppet | mpg123 have alot of issues with that |
| 17:19.03 | tessier | Sometimes I really hate cisco. |
| 17:19.09 | tessier | My Cisco 7960 says "Verifying load" and spins on it forever. :( |
| 17:19.13 | Darwin35 | mpg123-50r-pwhat ever is a pain |
| 17:19.23 | Darwin35 | it locks up |
| 17:19.27 | puppet | yeah it does |
| 17:19.34 | puppet | some time my music stop |
| 17:20.01 | Darwin35 | we need to find a replacement |
| 17:20.10 | Darwin35 | and dump mpg123 |
| 17:20.28 | file | make mpg123 |
| 17:20.31 | TheBear | puppet: are you using the agi festival-weather-script.pl ? |
| 17:20.32 | file | that's all you have to do in the asterisk directory |
| 17:20.36 | puppet | thebear: nope |
| 17:20.46 | puppet | file: thats not what we are discussing |
| 17:20.51 | Darwin35 | that ver does not work on fbsd |
| 17:20.54 | puppet | darwin35: a solution would maybe be an own player? |
| 17:21.05 | bjohnson | I have problems with mpg123 freezing up too |
| 17:21.10 | Darwin35 | so I use madplay |
| 17:21.10 | file | there's native music on hold too |
| 17:21.27 | puppet | file: "native" that uses mpg123 |
| 17:21.31 | file | no |
| 17:21.34 | puppet | yes |
| 17:21.35 | bjohnson | on fedora |
| 17:21.47 | file | no, native as in it reads in the format using the format modules |
| 17:21.49 | bjohnson | I want something that will play a stream |
| 17:21.51 | Darwin35 | what other whay is format_mp3 |
| 17:22.09 | puppet | ah format i thought he newt playmp3() |
| 17:22.21 | Darwin35 | that might work |
| 17:22.30 | puppet | playmp3 == mpg123 :/ |
| 17:22.32 | file | you can use playback, background, musiconhold, whatever.... with format_mp3 |
| 17:23.10 | puppet | darwin35: would you recommend to switch to madplay? |
| 17:23.20 | *** join/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 17:23.23 | Darwin35 | ok still no audio with 107 |
| 17:23.26 | Darwin35 | grrr |
| 17:23.41 | Darwin35 | hmm |
| 17:23.56 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 17:24.01 | Nohair | any idea about the transmit register |
| 17:24.04 | SexyKen | Hey guys -- I need some major help with Asterisk CDR. |
| 17:24.17 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 17:24.21 | *** part/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 17:24.33 | Hmmhesays | SexyKen is not a good nick |
| 17:24.40 | Hmmhesays | imho |
| 17:24.55 | Hmmhesays | for some reason it makes me think of "right said fred" |
| 17:25.43 | SexyKen | And you think 'Hmmhesays' is? Interesting. |
| 17:25.50 | spackle | puppet: I like madplay much better than mpg123 or whatever. |
| 17:26.01 | Hmmhesays | which in turn makes me think of that snl episode where conan obrien plays right said fred in a skit, that was funny |
| 17:26.04 | puppet | spackle: less lockups, less lagg? |
| 17:26.06 | Hmmhesays | so i guess it's alright |
| 17:26.14 | *** join/#asterisk ikey (ikey@220.226.14.149) |
| 17:26.34 | spackle | puppet: I think so, it just works. |
| 17:26.41 | Grooby | sigh |
| 17:26.43 | tessier | I really hope this 7960 hasn't turned itself into a doorstop. :( |
| 17:26.45 | tessier | It will take me weeks to get a replacement. |
| 17:26.47 | Grooby | #mythtv-users is so quiet |
| 17:26.53 | puppet | apt-get install madplay |
| 17:26.54 | puppet | ;D |
| 17:26.55 | tessier | And google search on "verifying load" turns up nothing |
| 17:26.57 | spackle | puppet: you can try it, if you don't like it, switch back. |
| 17:27.14 | Hmmhesays | i would think that gs would turn up loads of irrelevant information |
| 17:27.24 | Hmmhesays | mostly adult related |
| 17:27.32 | puppet | spackle: is ther ein wiki how to use it? |
| 17:27.34 | jakepdev | how many PRIs would you stick in a Dual Xeon 2.5Ghz * machine? |
| 17:27.40 | jakepdev | maximum? |
| 17:27.47 | tessier | jakepdev: Zero. |
| 17:27.49 | puppet | found ;D |
| 17:27.51 | Darwin35 | <PROTECTED> |
| 17:27.51 | Darwin35 | Darwin35 I dont hasve g729 on this box and it was looking for g729 |
| 17:28.02 | Darwin35 | I found my audio problem |
| 17:28.08 | Darwin35 | it all works now |
| 17:28.10 | p0lar | Anyone seen jer today? |
| 17:28.17 | p0lar | ~seen jer |
| 17:28.19 | jbot | p0lar: i haven't seen 'jer' |
| 17:28.19 | file | crazy coconuts |
| 17:28.26 | p0lar | ~seen jerjer |
| 17:28.27 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 12d 1h 9m 32s ago, saying: 'mrgoby: sure'. |
| 17:28.35 | TheBear | ~seen ManxPower |
| 17:28.36 | jbot | manxpower <~eric@dsl-208-164-150-160.datasync.com> was last seen on IRC in channel #asterisk, 3h 58m 33s ago, saying: 'I'll be in #asterisk-stable if people using 1.0.x have questions'. |
| 17:28.41 | puppet | spackle: does madplay takesstreams to? |
| 17:28.51 | CosmicRay | " Today, we were able to reach an astonishing 790 simultaneous audio playbacks. (sip to asterisk, any codec.) on a 350$ pc. Tests were done on we a pIV on 2.4ghz with 512mb ram." |
| 17:29.04 | p0lar | ~seen jerjer[mobile] |
| 17:29.05 | jbot | jerjer[mobile] <~jj@feth100-fw.fament.net> was last seen on IRC in channel #asterisk, 19h 1m 33s ago, saying: 'and we absolutely do not use the word unlimited'. |
| 17:29.15 | p0lar | closer |
| 17:29.23 | spackle | puppet: I haven't had time to figure that out yet. I use slimserver for MOH and want to use it with madplay if possible. |
| 17:29.36 | puppet | ok |
| 17:29.39 | SexyKen | Does anyone here know how to 'track' calls/join them together in the CDR so that we know certain 'calls' were connected to other ones? For instance, You dial 866 TESTING which enters you into an IVR. You select "1" for Test Further. Then you wait in a queue. When the queue clears, you are connected to an agent. -- That's a total of 3 calls that Asterisk logs into the CDR, but there's no way to know that the call is from the same origin |
| 17:30.30 | Corydon-w | Asterisk doesn't log 3 calls there -- only one |
| 17:30.40 | SexyKen | Corydon-w You're wrong. |
| 17:30.46 | SexyKen | It logs 3 calls. |
| 17:30.54 | SexyKen | For that scenario. |
| 17:30.58 | Nohair | pastebin.ca/7254 |
| 17:31.03 | Nohair | ANy one help?????\ |
| 17:31.20 | SexyKen | It logs one incoming to IAX, it logs one entering the queue, and it logs one connecting to the agent. |
| 17:32.17 | Darwin35 | mpg123 needs timing |
| 17:32.23 | *** join/#asterisk Dandan (dandan@234.88.149.195.in-addr.arpa.virt-ix.net) |
| 17:32.24 | SexyKen | Which makes sense, but there needs to be a way to link them together. Like a 'original-id' which should be a field in the db that is populated with the id of the FIRST call the was spawned for this group of calls. |
| 17:32.31 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 17:32.37 | puppet | lags as much :/ |
| 17:32.48 | Dandan | question: need someone with polycom phones experience, any1? |
| 17:32.58 | Nohair | darwin how do youi give mpg123 timing |
| 17:33.26 | Essobi | SexyKen In PBX land that's called a global-call-id.. the tranfers/spawns of an original call have a secondary field that has the original call leg in it. |
| 17:33.30 | Darwin35 | i believe zaptel should work |
| 17:33.37 | Zeeek | wtf???? http://wiki.e164.org/moin.cgi/TitleIndex |
| 17:33.45 | Essobi | err original global-call-id. |
| 17:33.53 | *** join/#asterisk __cmc__ (~cmc3c@209.101.178.227) |
| 17:33.59 | Nohair | darwin dont think i can run zaptel am using macos |
| 17:34.01 | Essobi | tess I think you have to upgrade the image. |
| 17:34.17 | SexyKen | Essobi - Do you know a way I can implement this into the CDR using the userfield? |
| 17:34.33 | Darwin35 | I dont use mpg123 due to issues |
| 17:34.45 | Nohair | darwin what do u use?\ |
| 17:34.47 | Darwin35 | but Iknow at 1 time moh and meetme needed zaptel |
| 17:34.52 | Darwin35 | madplayer |
| 17:34.56 | Dandan | Polycoms? anyone? |
| 17:35.19 | SexyKen | Polycom works okay for me. |
| 17:35.27 | Nohair | darwin what are you running asterisk on? |
| 17:35.31 | Essobi | SexyKen Hmm. should be able to use a per-channel variable, and store in the userfield in the dial-plan |
| 17:35.36 | Essobi | it's a dirty hack. |
| 17:35.42 | Dandan | SexyKen: can I priv with you? got some basic questions |
| 17:35.42 | Essobi | but it should work. |
| 17:35.43 | Darwin35 | fbsd 5.4 pre1 |
| 17:36.02 | Essobi | I want to get my hands on one of those T1 cards that have fbsd support |
| 17:36.07 | Essobi | and see if I can't get * to work with it |
| 17:36.09 | Darwin35 | the only mpg123 that works is in * |
| 17:36.24 | Darwin35 | you cd to your asterisk src and type make mpg123 |
| 17:36.39 | Darwin35 | but ti does not build on fbsd |
| 17:36.41 | SexyKen | Essobi, but the channel would change once the call comes in and switches from iax to sip...would there be a way to carry it over? |
| 17:36.44 | Darwin35 | so I use madplay |
| 17:36.46 | Nohair | darwin do you know what transmit_register is |
| 17:37.09 | Darwin35 | nope not delt with it |
| 17:37.45 | Darwin35 | thats for you to register with sip providers |
| 17:38.28 | __cmc__ | hi all ... wondering if there's anything like a consensus on which phones are the best for * |
| 17:38.46 | *** join/#asterisk xachen (justin@toto.citelnetworks.com) |
| 17:38.46 | Nohair | darwin i am registering but i keep getting a notice about it |
| 17:38.48 | SexyKen | Essboi, Still here? |
| 17:38.50 | tessier | __cmc__, No, there is not |
| 17:39.06 | xachen | Why is asterisk.or gnot working? |
| 17:39.14 | xachen | asteriski.org that is |
| 17:39.31 | xachen | oh now it works when I nag you guys haha |
| 17:39.41 | coppice | seen ~atacomm |
| 17:39.52 | puppet | trying native now |
| 17:40.01 | coppice | ~seen atacomm |
| 17:40.07 | jbot | atacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 36d 15h 43m 14s ago, saying: 'anyone want a IP 3000 conference phone? looking to replace ours with a IP 4000 model. Barely been used, in great condition.... looking for around $500'. |
| 17:40.18 | Dandan | SexyKen: i set up ftp server, phone logs in to it but i am curious how to upgrade firmware on it. I downloaded it to the ftp but the phone is unable to take it |
| 17:40.20 | Dandan | any ideas? |
| 17:40.52 | __cmc__ | I've heard good things about the Polycoms and Ciscos but also some discussion about how hard they are to get set up |
| 17:41.04 | lordcian | anyone here answer a quick AMPORTAL config question? |
| 17:41.20 | SexyKen | Dandan, You put bootrom.ld and sip.ld in root FTP directory? |
| 17:41.33 | Dandan | SexyKen: yup |
| 17:41.49 | Grooby | lordcian, you talking about AMP? |
| 17:42.08 | MicH323 | . |
| 17:42.12 | lordcian | grooby, yes. |
| 17:42.17 | SexyKen | •Dandan• What about sip.ver and bootrom.ver |
| 17:42.22 | *** join/#asterisk slav_jb (~k@pirus.securax.be) |
| 17:42.23 | Grooby | what's your question? |
| 17:42.37 | Grooby | no experts here but i'll try |
| 17:42.40 | Dandan | SexyKen: hold on, i goofed - chown ftp.ftp * -R helps... |
| 17:42.45 | lordcian | question about configuring '# Zap Channels = remove or add to this list as necessary' section of '/var/www/html/admin/retrieve_op_conf_from_mysql.pl' |
| 17:42.52 | lordcian | i have four Zap channels, they are 20-23 of my PRI, when i test they look like this in the console |
| 17:42.55 | puppet | hmmpg |
| 17:42.58 | lordcian | == Spawn extension (default, 2366, 3) exited non-zero on 'Zap/20-1' |
| 17:43.05 | lordcian | -- Executing Answer("Zap/20-1", "") in new stack |
| 17:43.10 | lordcian | how do i edit this line: $additional = "[Zap/1]\nPosition=1\nLabel=\"External 1\"\nExtension=-1\nIcon=3\n"; |
| 17:43.26 | SexyKen | Does anyone here know how to 'track' calls/join them together in the CDR so that we know certain 'calls' were connected to other ones? For instance, You dial 866 TESTING which enters you into an IVR. You select "1" for Test Further. Then you wait in a queue. When the queue clears, you are connected to an agent. -- That's a total of 3 calls that Asterisk logs into the CDR, but there's no way to know that the call is from the same origin |
| 17:43.27 | jakepdev | anyone use * with pri - is it difficult to configure? |
| 17:43.30 | TheBear | Ok I'm trying yet another weather script I can prove festival is working by dialing exten 860 exten => 860,1,Answer exten => 860,2,Festival (my name is) I hear my name is. Yet the script hangs and you hear nothing |
| 17:43.35 | lordcian | is it [Zap/20-1]\nPosition=1? |
| 17:43.38 | TheBear | anyone using the weather.sh script ? |
| 17:43.47 | puppet | Have anyone ever used native mp3 for MoH? |
| 17:44.20 | Grooby | anyone know? I don't actually use zap so I have never touch that side of the AMP |
| 17:44.25 | lordcian | or Zap/20-1\nPosition=20? |
| 17:44.30 | jakepdev | lordcian - is it easy to configure? |
| 17:44.41 | Dandan | SexyKen: worked, thanks :) |
| 17:44.51 | SexyKen | Yup. |
| 17:45.01 | lordcian | its been a little rough, but the docs are superior. mostly rough due to my lack of apache/mysql skills |
| 17:45.18 | Dandan | "Formatting file system. Please wait." |
| 17:45.24 | lordcian | this one Zap is fairly obscure, though. |
| 17:45.26 | Dandan | that's a good sign... :) |
| 17:45.31 | jakepdev | ok - tnx lordcian |
| 17:45.31 | Grooby | lordcian, what happens if you add the 20-1 and try it out |
| 17:45.33 | Grooby | ?? |
| 17:45.42 | Grooby | would you fry the hardware by accident? |
| 17:46.00 | jakepdev | what do you think abou the Sagoma Dual E1 card? |
| 17:46.04 | lordcian | i don't THINK so. :) |
| 17:46.06 | shido6 | boink |
| 17:46.18 | SexyKen | Damn! What happened to Essobi now...I wanted help :-( |
| 17:46.35 | lordcian | but, i don't actually understand the purpose of this config line, so i don't know what proper behavior would be... |
| 17:46.45 | SexyKen | shido, You must know a thing or two about CDR< no? |
| 17:47.03 | Bonbon | anyone know what the best rating engine is out there ? |
| 17:47.07 | Grooby | another good place to ask is the AMP sourceforge forum |
| 17:47.10 | Bonbon | or a good one |
| 17:47.14 | Grooby | I noticed they reply fairly quickly |
| 17:47.16 | Hmmhesays | rating engine? |
| 17:47.26 | jakepdev | if I'm using PRI - Zaptel or Sagoma? |
| 17:48.05 | *** join/#asterisk j_berry (~jberry@acs-24-154-150-174.zoominternet.net) |
| 17:48.27 | Bonbon | Hmmh: software which prices calls based on call plan etc. |
| 17:48.53 | coldfeet | Hi guys, anyone know howto leave a message for a particular login@context combo, when the prompt asks me all it wants is the mailbox number, but I have the same mailbox number in two diff contexts |
| 17:48.53 | shido6 | whats up , SexyKen ? |
| 17:48.56 | NetOfSickCoder | hi all |
| 17:49.07 | NetOfSickCoder | i'm new with asterisk |
| 17:49.29 | Grooby | brb |
| 17:49.46 | NetOfSickCoder | asterisk is installed, but there's several files |
| 17:49.55 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 17:50.27 | bannerman | This is odd, I was making calls a second ago, but now I get "Mar 11 09:50:10 NOTICE[31590]: channel.c:1352 ast_read: Dropping incompatible voice frame on IAX2/voipjet-3 of format g729 since our native format has changed to ulaw |
| 17:50.34 | bannerman | whenever I try to make a call |
| 17:50.40 | bannerman | spammed like a zillion times |
| 17:50.48 | bannerman | any idea what I screwed up? |
| 17:51.13 | *** join/#asterisk ^Fenris (~mazurbul@d3-99.rb.ot.centurytel.net) |
| 17:51.23 | jjg | anyone here know of a supermicro reseller in San Jose area? |
| 17:51.50 | file | jjg: did you come to the Digium booth @ VON? |
| 17:52.07 | TheBear | in a script how can I get the contents of a text file into a $variable to EXEC Festival $variable ? |
| 17:52.42 | xkev | can I bother someone to test dial a sip uri for me? |
| 17:52.47 | Qwell | $var = cat file; maybe |
| 17:52.51 | puppet | http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf < why doesnt that work with native? |
| 17:53.06 | file | puppet: look in the sample musiconhold.conf |
| 17:53.29 | file | there's... samples! |
| 17:54.03 | puppet | i have but it doesnt work :/ |
| 17:54.30 | jjg | file : no, I would never spend 200 bucks to go to an expo |
| 17:54.38 | ariel_ | bannerman, they might not be using g729 for there codec. set yours up to ulaw only and test again. |
| 17:54.41 | file | puppet: what did you do? |
| 17:54.46 | jjg | file : especially when no papers are being presented |
| 17:55.03 | jjg | file : although, i would have LIKED to gone |
| 17:55.09 | bannerman | Maybe I'm asking too much. I'm a newb. Whenever I try to call out, I get "Dropping incompatible voice frame on IAX2/voipjet-2 of format g729 since our native format has changed to ulaw". What's that mean? I don't think I've changed my config, not sure where to start. |
| 17:55.11 | Qwell | file: how much longer with VON? |
| 17:55.37 | file | VON is over |
| 17:55.41 | SexyKen | Does anyone here know how to 'track' calls/join them together in the CDR so that we know certain 'calls' were connected to other ones? For instance, You dial 866 TESTING which enters you into an IVR. You select "1" for Test Further. Then you wait in a queue. When the queue clears, you are connected to an agent. -- That's a total of 3 calls that Asterisk logs into the CDR, but there's no way to know that the call is from the same origin |
| 17:55.41 | Qwell | lame |
| 17:56.07 | ariel_ | bannerman, put in your iax.conf under your setup for voipjet disallow=all then allow=ulaw only that section. |
| 17:56.16 | puppet | file: http://pastebin.ca/7259 |
| 17:56.27 | bannerman | ariel_: that's what I have |
| 17:56.48 | file | did you put any files in the directory, does it exist, got format_mp3 installed if using MP3s? |
| 17:57.01 | puppet | format_mp3 is loaded |
| 17:57.23 | puppet | 1.1G /var/lib/asterisk/moh-native |
| 17:58.17 | Mw3 | hm |
| 17:58.20 | Mw3 | share it :) |
| 17:58.20 | ariel_ | bannerman, what server are you using there ny ones or california one? |
| 17:58.51 | bannerman | voipjet only has one, as far as I know, and it's east coast.. not sure where exactly |
| 17:58.59 | Zeeek | bannerman no |
| 17:59.04 | Zeeek | there are three of four |
| 17:59.08 | bannerman | oh |
| 17:59.16 | file | puppet: restarted asterisk? |
| 17:59.19 | Zeeek | they were mailed to users a few days ago |
| 17:59.20 | puppet | file: yeah |
| 17:59.22 | Darwin35 | coolio |
| 17:59.31 | bannerman | I signed up this morning, didn't get any email :) |
| 17:59.32 | Darwin35 | broadvoice works again |
| 17:59.54 | Zeeek | bannerman then it should be in the page about setup |
| 17:59.57 | Goshen | Darwin35: how does outgoing CallerID work on Broadvoice? |
| 18:00.08 | Goshen | I want to control my outgoing caller ID |
| 18:00.13 | puppet | file: all looks right? |
| 18:00.21 | Darwin35 | setcid is not working but it is sending the nmbr |
| 18:00.23 | Goshen | the comment on the wiki said that they show private number no matter what |
| 18:00.38 | Goshen | so it gives out your number when you dial out? but not the name? |
| 18:00.42 | bannerman | oh, and there it is |
| 18:00.43 | bannerman | thanks man |
| 18:00.49 | Darwin35 | yeah |
| 18:01.04 | Darwin35 | I wish they would fix the setcidname |
| 18:01.11 | Darwin35 | I want to send my company name |
| 18:01.19 | Zeeek | setcidname(GW Bush) |
| 18:01.20 | Goshen | the comment on the wiki that someone left today says he has been on the phone and the people on the phone have no idea what he is talking baout |
| 18:01.33 | Goshen | and he made them look in to it, and they set the name manually, but it still didn't work |
| 18:01.47 | Goshen | yea, I want to send my company name too |
| 18:01.57 | Goshen | I haven't signed up with them yet |
| 18:02.30 | Goshen | I guess you could just send outgoing calls over NuFone |
| 18:02.30 | Darwin35 | that cost |
| 18:02.34 | Goshen | sure a little bit |
| 18:02.35 | Darwin35 | I have unlimited outgoing with BV |
| 18:02.38 | Goshen | right |
| 18:02.47 | Darwin35 | 19.95 a month |
| 18:02.53 | Darwin35 | unlimited |
| 18:02.58 | Goshen | ok, I will go get signed up with them, I want them to port three of my numbers |
| 18:03.10 | TheBear | exten => 860,2,Festival(mary has a little lamb) works yet exten => 855,2,Festival(online forecast for) does not, any reasons why ? |
| 18:03.31 | Darwin35 | the bear you touched it |
| 18:03.35 | Darwin35 | ducking |
| 18:03.35 | puppet | thebear: cause the festival modules is crazy |
| 18:03.47 | puppet | ;D |
| 18:03.51 | Darwin35 | wait is it not bear season yet |
| 18:03.57 | Goshen | Darwin35: so is BV all fixed up now? |
| 18:04.01 | Goshen | did you have to apply your patch? |
| 18:04.14 | Qwell | BV hates *, heh |
| 18:04.17 | Goshen | does 1.0.6 need to be patched? |
| 18:04.19 | Darwin35 | the patch is in 1.0.7 |
| 18:04.23 | Inv_arp | BV always werked for me |
| 18:04.26 | TheBear | puppet: ok will AGI do text to speech ? |
| 18:04.30 | Goshen | ok 1.0.7 is released? |
| 18:04.41 | file | read the darn topic you crazy dingbats |
| 18:04.49 | puppet | thebear: yes read wiki |
| 18:04.57 | Goshen | topic says RC |
| 18:04.58 | file | 1.0.7 RC - bug #3746 |
| 18:04.59 | JunK-Y | Goshen: read topic. |
| 18:05.01 | JunK-Y | :) |
| 18:05.02 | file | Release Candidate |
| 18:05.04 | file | it means test it |
| 18:05.06 | Darwin35 | goshen I can send you the patched chan_file |
| 18:05.06 | file | use it |
| 18:05.08 | file | comment on it |
| 18:05.23 | Goshen | where is the bug tracker? |
| 18:05.25 | JunK-Y | file: back to NB ? |
| 18:05.37 | JunK-Y | Goshen: http://bugs.digium.com |
| 18:05.42 | Goshen | thanks |
| 18:06.20 | coldfeet | Guys when the Voicemailmain prompt plays I seem to always miss the first part, and I am not sure if this is because I am connecting via SER --> asterisk |
| 18:06.34 | Goshen | Is the VOIP adapter BroadVoice sends locked? |
| 18:07.16 | TheBear | puppet: care to help where all I'm seeing in wiki is festival |
| 18:07.28 | antifuchs | coldfeet: happens to my phones too, and I'm using asterisk as the SIP server |
| 18:07.38 | puppet | thebear: and there is a perlscript for agi |
| 18:07.47 | Goshen | Darwin35: give me a referral link to broadvoice so you get credit |
| 18:07.55 | puppet | coldfeet: add a wait(1) before calling it |
| 18:08.20 | coldfeet | trying it now puppet |
| 18:08.27 | puppet | coldfeet: it works |
| 18:08.32 | Darwin35 | 412-28-1009 |
| 18:08.36 | Darwin35 | 238 |
| 18:10.24 | *** join/#asterisk criptos (~criptos@dsl-200-67-175-109.prod-empresarial.com.mx) |
| 18:10.29 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 18:10.37 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 18:10.45 | coldfeet | puppet no luck even tried wait 5 |
| 18:10.48 | criptos | What software client with IAX support for windows can I use? And is reliable? |
| 18:11.09 | Goshen | Criptos: there are several... |
| 18:11.10 | Goshen | ~firefly |
| 18:11.11 | jbot | methinks firefly is http://virbiage.com/firefly/download/firefly-thirdparty.exe |
| 18:11.16 | coldfeet | it still chops of the first part, of VoiceMailMain |
| 18:11.21 | Zeeek | IAXCOMM |
| 18:11.35 | Zeeek | IAXPhone, DIAX |
| 18:11.35 | *** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com) |
| 18:11.41 | criptos | Zeeek Iaxcomm, :) nice. |
| 18:11.43 | criptos | thanks! |
| 18:12.06 | Zeeek | I should keep all those links in one place |
| 18:12.17 | Zeeek | http://iaxclient.sourceforge.net/iaxcomm/ |
| 18:12.30 | puppet | coldfeet: it shall work :) |
| 18:12.33 | Zeeek | http://www.sokol-associates.com/IaxPhone.htm |
| 18:12.36 | coldfeet | it doesnt :-( |
| 18:12.45 | puppet | coldfeet: paste your dialplain in pastebin |
| 18:12.49 | puppet | ~pastebin |
| 18:12.50 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 18:12.52 | coldfeet | tried increasing it, but even if i go from 5-10, it still chops of just that part |
| 18:12.54 | NetOfSickCoder | howto up my asterisk sip server :( ? |
| 18:12.59 | Zeeek | http://www.laser.com/dante/diax/diax.html |
| 18:14.15 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 18:14.31 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 18:14.54 | coldfeet | puppet http://pastebin.ca/7261 |
| 18:15.23 | puppet | coldfeet: why are you making it as a macro? |
| 18:15.46 | coldfeet | because I have other macros there and its cleaner eventually |
| 18:15.51 | bannerman | Switching to the left coast server made it work. So.. what was causing my problem? Am I going to run into this again? |
| 18:16.14 | puppet | i would have put it as one |
| 18:16.31 | coldfeet | yeah I did initially but I have other contexts and this is a test one for voicemail |
| 18:16.46 | coldfeet | would it make a diff to the call |
| 18:16.50 | Inv_arp | bannerman: which prodider? |
| 18:16.57 | bannerman | voipjet |
| 18:17.25 | file[SFO] | gone to San Francisco, bbl |
| 18:17.27 | Inv_arp | bannerman: hmm do u have the 3 ip's i use the east coast one... lemme ping to see who is faster |
| 18:17.28 | coldfeet | also on the sam note, when I dial into my mailbox , and leave a message, it asks for mailbox number..how do I set the mailbox context if I have the same mailbox number for two contexts |
| 18:17.42 | bannerman | Inv_arp I have the IPs, yeah |
| 18:17.54 | bannerman | Inv_arp I just don't understand what caused the error I was getting " |
| 18:17.54 | bannerman | *CLI> -- Acceptin |
| 18:17.56 | bannerman | er.. |
| 18:18.01 | bannerman | some error heh |
| 18:18.19 | puppet | omfg lagg |
| 18:18.22 | Inv_arp | what are the ip's or where do i find them on their site |
| 18:18.43 | bannerman | Primary East Coast Server: 216.118.117.46 |
| 18:18.43 | bannerman | Backup East Coast Server: 66.246.220.19 |
| 18:18.43 | bannerman | Primary West Coast Server: 69.25.60.30 |
| 18:18.51 | bannerman | so is latency the cause of my problem? |
| 18:19.00 | bannerman | or a bad configuration for codecs |
| 18:19.11 | SexyKen | Does anyone here know how to 'track' calls/join them together in the CDR so that we know certain 'calls' were connected to other ones? For instance, You dial 866 TESTING which enters you into an IVR. You select "1" for Test Further. Then you wait in a queue. When the queue clears, you are connected to an agent. -- That's a total of 3 calls that Asterisk logs into the CDR, but there's no way to know that the call is from the same origin |
| 18:19.18 | bannerman | do I need to allow a low quality codec on my phone and asterisk server? |
| 18:19.35 | coldfeet | puppet it seems to be a problem at the client side, if I try with xlite it seems to work |
| 18:19.51 | Inv_arp | bannerman: yep i use primary east 64 bytes from 216.118.117.46: icmp_seq=1 ttl=53 time=71.4 ms |
| 18:20.02 | Inv_arp | gsm codec |
| 18:20.28 | bannerman | gsm doesn't sound nearly as good as ulaw |
| 18:20.38 | bannerman | should I enable gsm as a backup? |
| 18:20.42 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 18:21.04 | Inv_arp | bannerman: lol for a normal convo i cant tell diff |
| 18:21.49 | *** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 18:22.19 | bannerman | I can. It may be something other than the difference between codecs, but GSM sounds fuzzy and ulaw sounds flawless. |
| 18:22.40 | Skysky | may i get some help? i received a error msg when using festival. "client(1) Fri Mar 11 13:20:14 2005 : rejected from PBX not in access list" |
| 18:22.51 | bannerman | my ping to the pri east server is about 100, west coast server is 30... wahoo :) |
| 18:23.36 | coldfeet | Ooh, format changed from unknown to ulaw.....how do I get rid of this, do I set the first codec on client to ulaw |
| 18:23.53 | *** join/#asterisk _omer (dfsdf@202.147.174.177) |
| 18:23.58 | _omer | hi |
| 18:24.04 | SexyKen | Inv_arp, Hey remember my issue with 'call tracking'? |
| 18:24.18 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 18:24.21 | _omer | Can I listen to the conversations in asterisk?? I mean stealth listening??? |
| 18:24.24 | Skysky | coldfeet: disallow=all then allow=ulaw |
| 18:24.27 | PTG123 | Anyone know how to get the average hold time for a queue? |
| 18:24.48 | Qwell | PTG123: ever get that SQL problem fixed last night? |
| 18:25.08 | Inv_arp | SexyKen: still no luck ehh |
| 18:25.15 | SexyKen | Inv_arp, Not at all. |
| 18:25.17 | _omer | <PROTECTED> |
| 18:25.20 | PTG123 | qwell: which one? |
| 18:25.29 | SexyKen | Inv_arp - I just can't find a way to track the calls from where they were originated |
| 18:25.31 | Qwell | I forget now. |
| 18:25.34 | PTG123 | heh |
| 18:25.43 | PTG123 | well imanaged to sync up most of the cdr records i think |
| 18:25.46 | Qwell | something about grouping functions vs non grouping |
| 18:25.57 | PTG123 | oh yah |
| 18:26.01 | PTG123 | it was stupid :) |
| 18:26.55 | bannerman | I'm real concerned about this. The following error "Dropping incompatible voice frame on IAX2/voipjet-3 of format g729 since our native format has changed to ulaw" came up randomly, with no config change, until I switched to a voipjet server closer to me. What causes that, and how can I avoid it? |
| 18:27.08 | Inv_arp | SexyKen: the only thing i can recommend, if by default cdr doesnt stamp an id or somethin per call is to use AGI with itto place a unique identifier per call |
| 18:27.23 | _omer | anybody????? |
| 18:27.27 | _omer | <PROTECTED> |
| 18:27.29 | *** join/#asterisk mesi (~player@dsl-082-083-128-010.arcor-ip.net) |
| 18:27.38 | Qwell | _omer: no need to repeat |
| 18:27.41 | mesi | BV is Broadvoice? |
| 18:27.42 | Qwell | 3 times in 3 minutes |
| 18:27.45 | Qwell | mes: yes |
| 18:27.49 | mesi | ok. |
| 18:28.01 | Qwell | _omer: Its possible, yes. I doubt you'll get much help if you keep asking though. |
| 18:28.02 | SexyKen | PTG123, I think I found your average queue time while agents are 'busy' |
| 18:28.07 | *** join/#asterisk tekjacob (~tekjacob@c2.efb7d1.client.atlantech.net) |
| 18:28.14 | SexyKen | Or at least how to get the records. |
| 18:28.34 | tekjacob | Hey all, is it acceptable to ask Flash Operator Panel questions here? |
| 18:28.50 | Inv_arp | _omer: doesnt seem hard... think convos can be recorded not sure on live monitoring |
| 18:29.15 | Inv_arp | tekjacob: might be on way to find out |
| 18:29.21 | Inv_arp | s/on/one |
| 18:29.31 | _omer | alright.... |
| 18:29.33 | Hmmhesays | wtf mpg123 is taking up 100% of my processor |
| 18:30.03 | denon | crontab killall -9 :) |
| 18:30.14 | _omer | Qwell: are you sure about live monitoring/listening??? that its possible? |
| 18:30.17 | Darwin35 | get madplay |
| 18:30.22 | Darwin35 | drop mpg123 |
| 18:30.28 | Inv_arp | Hmmhesays: pkill -kill mpg123 |
| 18:30.28 | Qwell | _omer: likely |
| 18:30.37 | Inv_arp | Damin: madplay? |
| 18:30.46 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 18:30.53 | Hmmhesays | i find it odd that playing 2 on hold music channels should be taking 100% of the processor |
| 18:31.01 | Darwin35 | madplay is a mp3 player |
| 18:31.03 | denon | or quit trying to run it on a 386 :) |
| 18:31.10 | tekjacob | alright... I have a odd setup... I have calls coming in via PRI to * box a which sends them via IAX to asterisk box b where all the clients are registered. I want to be able to click on a call in FOP and drop it on a speed dial button to send it back across the IAX to *a and back out the PRI |
| 18:31.26 | tekjacob | but I seem to end up in random places in my dialing plan |
| 18:32.28 | Inv_arp | heh i run * on 350mhz mpg123 runs fine not even 1% of cpu |
| 18:32.31 | *** part/#asterisk jjg (~jjg@63.166.156.128) |
| 18:32.32 | Goshen | I like this web based configuration for broadvoice features..that is nice |
| 18:33.09 | *** part/#asterisk __cmc__ (~cmc3c@209.101.178.227) |
| 18:33.15 | *** part/#asterisk criptos (~criptos@dsl-200-67-175-109.prod-empresarial.com.mx) |
| 18:33.46 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 18:34.14 | *** join/#asterisk lidl (~little@81-208-60-192.fastres.net) |
| 18:35.04 | Inv_arp | tekjacob: FOP? |
| 18:35.16 | tekjacob | "Flash Operator Panle" |
| 18:35.17 | lidl | is it possible to have 2 pstn lines entering a tdm400 card answering at the same number? or is it compulsory to have isdn for the two channels? |
| 18:35.18 | *** join/#asterisk jmhunter (~jmhunter@wire3-148.razzolink.com) |
| 18:35.18 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 18:35.20 | tekjacob | Panel |
| 18:35.28 | jontow | http://mno.bsd.st/~jontow/2005-03-11-systat.tgz |
| 18:35.33 | jontow | my experiement with AGI :) |
| 18:35.37 | lesouvage | I like to use a x100p or alternative to my laptop. Is there a cheaper way to this then this external PCI slot box? http://www.mycableshop.com/sku/PCI-CB1H.htm |
| 18:35.40 | jontow | (experiment) |
| 18:35.44 | Qwell | lidl: Do you have two lines? |
| 18:35.49 | *** join/#asterisk afe ([PRosOd3P+@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 18:36.30 | lidl | Qwell, i don't have any atm :). I'm going to make a contract soon anyway, so i was wondering about this question |
| 18:36.53 | Qwell | lidl: If I'm understanding the question right(you may wish to rephrase it), then sure, why wouldn't you be able to? |
| 18:37.09 | *** join/#asterisk [Latre] (~latre@148.233.19.133) |
| 18:37.23 | Darwin35 | Kiss me Blarney stone |
| 18:37.25 | jmhunter | can anyone recomend an affordable tiny PC solution? |
| 18:37.39 | *** part/#asterisk slav_jb (~k@pirus.securax.be) |
| 18:37.42 | outtolunc | winterm 50 <G> |
| 18:37.42 | Goshen | lidl: you want the second PSTN to ring into the card if the first one is busy? |
| 18:37.54 | lidl | Goshen, yes |
| 18:37.55 | [Latre] | someone knows people that sell x100p in Mexico??? |
| 18:37.57 | Inv_arp | jmhunter: shuttle setups are tiny |
| 18:38.03 | Goshen | lidl: call forward busy line... |
| 18:38.09 | TheBear | anyone on the lists.digium.com admin staff, I've tried twice to subscribe to asterisk-users and no sucess ?? |
| 18:38.14 | Goshen | its a feature for residential customers only costs 35 cetns |
| 18:38.16 | Goshen | cents |
| 18:38.22 | Goshen | and for business is 4.00 |
| 18:38.25 | Goshen | a month |
| 18:39.02 | Goshen | Lesouvage: don't use the pci card for laptop, get an ATA |
| 18:39.05 | ariel_ | jmhunter, how tiny? |
| 18:39.22 | Qwell | oh, I see... Each line would have a different DID. If line 1 is busy, forward to line 2's DID? |
| 18:39.36 | ariel_ | lesouvage, you can get a sipura 3000 it gives you 1 fxo and 1 fxs port. |
| 18:39.53 | Qwell | I guess you can't get 2 residential lines, with only 1 number, huh? |
| 18:40.20 | lidl | Goshen, is it then possible to have an ADSL on one of the two lines? or does it "conflict" with asterisk? |
| 18:40.32 | Qwell | lidl: as long as you have all the filters and stuff |
| 18:40.40 | Qwell | if a phone works, * can work |
| 18:40.49 | lidl | thx :) |
| 18:40.50 | ariel_ | adsl does not conflict just use the filter on it. |
| 18:41.01 | Darwin35 | ok back live from Pittsburgh its the Darwin hour |
| 18:41.18 | ariel_ | Pittsburgh is it cold there? |
| 18:41.29 | [Latre] | what chispets are good for x100p clon? |
| 18:41.45 | [Latre] | 537PU and PG ? and one else? |
| 18:41.45 | ariel_ | [Latre], only the intel work |
| 18:42.11 | [Latre] | ariel_: you know where i found that modems? |
| 18:42.13 | Darwin35 | Allegheny County Airport |
| 18:42.13 | Darwin35 | Last updated on March 11, 2005 12:53 pm |
| 18:42.13 | Darwin35 | Broken clouds at 800 feet |
| 18:42.13 | Darwin35 | Overcast clouds at 1300 feet |
| 18:42.13 | Darwin35 | Unknown Precipitation |
| 18:42.13 | Darwin35 | Mist |
| 18:42.18 | Darwin35 | Temperature: |
| 18:42.19 | Darwin35 | 35.1?F |
| 18:42.22 | Darwin35 | <PROTECTED> |
| 18:42.23 | Darwin35 | Dew Point: |
| 18:42.25 | Darwin35 | 33.1?F |
| 18:42.27 | Darwin35 | Air Pressure: |
| 18:42.30 | Darwin35 | 29.40" Hg |
| 18:42.31 | Darwin35 | <PROTECTED> |
| 18:42.34 | Darwin35 | Rel. Humidity: |
| 18:42.35 | Darwin35 | 92.4% |
| 18:42.37 | Darwin35 | Wind Speed: |
| 18:42.40 | Darwin35 | 12 MPH WSW |
| 18:42.41 | Darwin35 | <PROTECTED> |
| 18:42.43 | Darwin35 | Wind Chill: |
| 18:42.47 | Qwell | could have just said, "yeah, its nice and cool" |
| 18:42.47 | Darwin35 | 26.3?F |
| 18:42.50 | Darwin35 | Sunrise: |
| 18:42.51 | Darwin35 | 06:36 am |
| 18:42.53 | Darwin35 | <PROTECTED> |
| 18:42.55 | Darwin35 | Sunset: |
| 18:42.57 | Darwin35 | 06:22 pm |
| 18:43.03 | Darwin35 | sorry |
| 18:43.05 | Darwin35 | damn that script |
| 18:43.06 | dsfr | hah |
| 18:43.07 | Darwin35 | I have to fix it |
| 18:43.09 | Darwin35 | its not suppost to give all that info |
| 18:43.25 | [Latre] | ariel_: you know where i found that modems? |
| 18:43.38 | Darwin35 | what modem |
| 18:43.40 | ariel_ | [Latre], I don't use the clones |
| 18:43.49 | [Latre] | 537PU and PG intel |
| 18:43.57 | [Latre] | x100p clon |
| 18:44.07 | Darwin35 | google it |
| 18:44.08 | Qwell | If you insist on buying a clone, look on ebay |
| 18:44.08 | [Latre] | or x100p cheaper |
| 18:44.08 | ariel_ | [Latre], most here will not try to help with you getting the clone x100p boards. |
| 18:44.17 | spackle | ~weather KDSM |
| 18:44.21 | coldfeet | how do u customise Comedian mail , i.e change the prompts, is it just replacing the sound files |
| 18:44.23 | Darwin35 | there is a site that has them for 20 bucks |
| 18:44.28 | ariel_ | ~weather ktmb |
| 18:44.30 | boch | [Latre] ebay |
| 18:44.41 | KalD|Work | ~weather KANC |
| 18:44.47 | KalD|Work | lol |
| 18:45.04 | ariel_ | 75 outside great |
| 18:45.22 | lidl | I've just called my national telephony operator |
| 18:45.40 | lidl | they told me it's not possible to have 2 calls on the same number |
| 18:45.51 | Qwell | You didn't ask right. :p |
| 18:45.57 | KalD|Work | ~weather PAOM |
| 18:46.04 | Qwell | Did you ask about the busy call forwarding? |
| 18:46.19 | *** join/#asterisk Uajal (~icechat5@ool-182e86f3.dyn.optonline.net) |
| 18:46.24 | Dandan | ~weather USCT0139 |
| 18:46.29 | lidl | i would need two different pstn lines. if the first is busy, then the call is forwarded to the other line (which has a different public number) |
| 18:46.30 | Darwin35 | its in the wiki |
| 18:46.48 | Qwell | lidl: That seems like a sane solution. |
| 18:47.00 | Qwell | or pay 10x as much for a BRI |
| 18:47.04 | Dandan | ~weather 0139 |
| 18:47.19 | lidl | Qwell, in fact i would like to avoid isdn |
| 18:47.19 | SexyKen | Does anyone here know how to 'track' calls/join them together in the CDR so that we know certain 'calls' were connected to other ones? For instance, You dial 866 TESTING which enters you into an IVR. You select "1" for Test Further. Then you wait in a queue. When the queue clears, you are connected to an agent. -- That's a total of 3 calls that Asterisk logs into the CDR, but there's no way to know that the call is from the same origin |
| 18:47.22 | Darwin35 | its called rollover |
| 18:47.43 | Dandan | ~weather KHFD |
| 18:47.48 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 18:48.20 | *** join/#asterisk ZOP (NachoGod@ZOP.sustaining.supporter.pdpc) |
| 18:49.06 | Uajal | Hi guys, how should I call external numbers with IAX (with sip it was SIP/16465157654@sip.broadvoice.com). Now I am at voicepuls. |
| 18:49.29 | *** part/#asterisk [Latre] (~latre@148.233.19.133) |
| 18:49.32 | Darwin35 | iax2/ |
| 18:49.42 | lidl | what if i have a fax? I mean someone faxes me, will asterisk be able to recognise it as a fax call and route it to the fax attached for example to the FXS port (once configured)? |
| 18:49.46 | `Sauron | iax.. what darwin said |
| 18:49.46 | ZOP | Darwin35: no i hate the inspectors more when they're being anal. we're being told we can not have ANY extension cords (including breakered six outlet cords) plugged into battery backups. |
| 18:50.29 | Darwin35 | wow thats anal |
| 18:50.30 | TheBear | lidl: yes |
| 18:50.36 | Uajal | Darwin35:iax2/ ... and what then. I tried but did something wrong |
| 18:50.39 | ZOP | lidl: in your incoming context have a fax extension (literally)...tho that may only work on ZAP channels. |
| 18:50.44 | Darwin35 | unless it a small ups and ment for 1 to 2 machines |
| 18:50.57 | ZOP | Darwin35: no we're talking APC1400+es |
| 18:51.04 | ZOP | all 1400VA and larger |
| 18:51.09 | Darwin35 | ahh ok |
| 18:51.10 | ZOP | we were nicked twice on that. |
| 18:51.12 | Darwin35 | thats anal |
| 18:51.14 | `Sauron | yeah |
| 18:51.19 | `Sauron | get power distribution things |
| 18:51.22 | `Sauron | 1U 8 outlet things |
| 18:51.26 | `Sauron | MMM..... |
| 18:51.33 | ZOP | we do have a little 750VA running two machines with extensions but he didnt' see that heh |
| 18:51.40 | ZOP | no i'm talking in our ofice |
| 18:51.42 | ZOP | *office |
| 18:52.15 | spackle | ~office |
| 18:52.21 | ZOP | we use a Symmetra LXes upstairs in facilities area, those are hardwired so there is no question you can plug PDUs and such into those monsters. |
| 18:52.55 | lidl | finally, I'd like to avoid using WiFi phone as well as DECT phone. I'd like to use pstn cordless phone. May I plug the pstn to iaxy and connect iaxy to the local network and treat it as an ip phone? |
| 18:53.26 | lidl | localnet->iaxy->pstn cordless phone |
| 18:53.36 | lidl | would this be sane? |
| 18:54.06 | lidl | outside_world->asterisk->localnet->iaxy->pstn cordless phone |
| 18:54.07 | *** join/#asterisk ws9455 (~ws9455@adsl-68-94-48-58.dsl.rcsntx.swbell.net) |
| 18:54.20 | Dandan | SexyKen: u still here? |
| 18:54.58 | Katty | hmmmm |
| 18:55.20 | `Sauron | lidl: works fine |
| 18:55.30 | Uajal | iax2/ what then? I tried IAX2/16465157654. Result: no such host 16465157654. Unable to request channel IAX2/16465157654. |
| 18:55.45 | ariel_ | hello Katty, so have you gotten your asterisk system up and running yet? |
| 18:56.07 | `Sauron | iax2/phonenumber@iaxprovider |
| 18:56.20 | Katty | ariel_: moment, on phone |
| 18:56.22 | `Sauron | iax2/blahblah@fwd |
| 18:56.24 | `Sauron | etc |
| 18:56.44 | TheBear | which file changes the voicemail "Comedian Mail, mailbox" ? |
| 18:58.11 | ariel_ | TheBear, I think it's vm-login.gsm |
| 18:58.16 | ariel_ | but don't remember |
| 18:58.50 | lidl | well, my national phone operator charges more for two analog phone lines than for an ISDN line (with two channels) |
| 18:58.55 | bannerman | I'm real concerned about this. The following error "Dropping incompatible voice frame on IAX2/voipjet-3 of format g729 since our native format has changed to ulaw" came up randomly, with no config change, until I switched to a voipjet server closer to me. I need to figure out what causes that, and how can I avoid it. Do I need to have GSM as a backup codec? |
| 18:58.55 | ariel_ | dial,iax2/username:password@ipaddress or context/${EXTEN} |
| 18:59.09 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 18:59.13 | SexyKen | Dandan, Yes |
| 18:59.27 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 18:59.29 | Dandan | SexyKen: can you paste any .cfg file into a pastebin or sth? |
| 18:59.38 | SexyKen | What does 'busy' mean in the disposition field of Asterisk CDR? |
| 18:59.38 | Dandan | i am having problems with this phone... |
| 18:59.49 | SexyKen | Dandan, what's the problem |
| 18:59.52 | SexyKen | And what SP is it? |
| 18:59.59 | Dandan | IP 300... |
| 19:00.03 | Dandan | can't register |
| 19:00.09 | ariel_ | bannerman, give them an email and ask. They has some problems with there servers yesterday. I just made a call via there server now and it worked. |
| 19:00.12 | Darwin35 | ooops |
| 19:00.17 | SexyKen | Show me your Asterisk error |
| 19:00.18 | Dandan | we are testing it to deploy about 70 of them... |
| 19:00.27 | Darwin35 | pastbin it |
| 19:00.29 | Dandan | if i manage this one to work properly... |
| 19:00.32 | Darwin35 | ~pastebin |
| 19:00.33 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 19:00.54 | Darwin35 | ~jbot |
| 19:00.56 | jbot | jbot is probably the shipboard computer, but you may call me eddie if it helps you relax |
| 19:01.13 | Darwin35 | ~darwin |
| 19:01.14 | jbot | rumour has it, darwin is an apple-sourced os based on BSD, also see MacOS X or the APSL. |
| 19:01.21 | tzafrir_laptop | There's a basic point I don't exactly understand. Suppose I set my * to listen on SIP or IAX an there is an incomming connection. How does * know which extension from sip/iax.conf to use for this connection? |
| 19:01.47 | Darwin35 | thats in extensions.conf |
| 19:01.52 | SexyKen | Dandan, what is the error |
| 19:02.10 | Dandan | beep beep beep beep... |
| 19:02.14 | Dandan | :) |
| 19:02.20 | ariel_ | tzafrir_laptop, its up to the context which you setup via the sip.conf or iax.conf then what it matches in the extensions.conf. |
| 19:02.22 | Dandan | asterisk doesn't even notice it... |
| 19:02.29 | Darwin35 | or at the end of you register line yo hae a /{extension} like /s and it is read from the exteensions file |
| 19:02.32 | Dandan | I might not configured it correctly |
| 19:02.40 | Dandan | that's y i would like to look at yours |
| 19:02.47 | Darwin35 | and thhen goes to where it is pointed |
| 19:02.48 | *** join/#asterisk tomas- (~tomas@pcp0010083068pcs.frnklv01.nj.comcast.net) |
| 19:02.52 | tzafrir_laptop | ariel_, but how does * know which sip.conf context to use? |
| 19:02.54 | SexyKen | Dandan -- show me your conf and I'll tell you if it's wrong. |
| 19:03.07 | Dandan | SexyKen: hold on :) |
| 19:03.22 | ariel_ | tzafrir_laptop, sip devices have to send info like login and such. |
| 19:03.38 | Darwin35 | I use exten # and not names on my sip.conf extensions |
| 19:03.49 | bannerman | ariel_: thanks a ton |
| 19:03.50 | Darwin35 | that way macro-allexten workls |
| 19:04.05 | Bonbon | anyone use a rating engine? |
| 19:04.11 | tzafrir_laptop | ariel_, and if I want to allow people (strangers) to call me through SIP or IAX, how do I do that? |
| 19:04.24 | xkev | ..until the next position announce, or whatever |
| 19:04.32 | xkev | wrong window |
| 19:04.33 | ariel_ | tzafrir_laptop, your asking for trouble there. |
| 19:04.51 | Darwin35 | grrr |
| 19:04.54 | ariel_ | but if you want someone to call you via sip lets say they can dial your exten like 100@youripaddress. |
| 19:04.59 | tomas- | is it possible to have an asterisk pbx that takes a single phone line and can simultaneously cause calls to ring on multiple phones? would this show up on a different button on the phone, meaning youd be able to identify what number was called in to? |
| 19:05.13 | Darwin35 | life is like a tootsie pop |
| 19:05.22 | tzafrir_laptop | ariel_, why? anybody in the world can call me by phone. why not also by SIP/IAX ? |
| 19:05.41 | *** join/#asterisk _santiago_ (~santiago@63.245.86.95) |
| 19:05.48 | Darwin35 | you suck and suck and suck and then its just gone |
| 19:05.55 | ariel_ | tzafrir_laptop, they can what is your servers ip address and your extension? |
| 19:06.02 | `Sauron | you can call me by dialing my name@mydomain.com |
| 19:06.15 | `Sauron | well |
| 19:06.24 | `Sauron | apparently, it's name@sip.domain.com |
| 19:06.43 | Darwin35 | ? |
| 19:06.46 | `Sauron | since the SRV stuff doesn't want to work, apparently. |
| 19:06.47 | ariel_ | it's extensionnumber@ipaddress or domain address. |
| 19:07.15 | `Sauron | ariel: you can make an extension be name@ |
| 19:07.30 | tzafrir_laptop | ariel_, yes, this is what I want. Does it require setting an incoming extension? |
| 19:07.36 | ariel_ | yes but why? |
| 19:07.41 | `Sauron | why not |
| 19:07.51 | ariel_ | tzafrir_laptop, yes |
| 19:07.51 | tzafrir_laptop | If not: how does * know to which context to send those calls? |
| 19:08.05 | `Sauron | [default] |
| 19:08.05 | `Sauron | exten => dominic,1,Goto(from-sip|s|1) |
| 19:08.18 | Uajal | Could you give me literal sample of IAX or IAX2 outbound call |
| 19:08.22 | `Sauron | which dumps it into the dialplan for all my other inbound sip calls... |
| 19:08.27 | ariel_ | sauron you can but allot of devices can't dial names |
| 19:08.40 | tzafrir_laptop | `Sauron, in sip.conf |
| 19:08.49 | `Sauron | tzafrir: in extensions.conf |
| 19:09.26 | `Sauron | Uajal: exten => _1393.,3,Dial(${FWDGW}/${EXTEN:3}|60|r) |
| 19:09.32 | ariel_ | tzafrir_laptop, you need to setup a device to register to your asterisk box. then give it an extension. or maybe a zap port. |
| 19:09.58 | tzafrir_laptop | ariel_, must they register? |
| 19:10.16 | ariel_ | tzafrir_laptop, yes |
| 19:10.17 | *** join/#asterisk tomas- (~tomas@pcp0010083068pcs.frnklv01.nj.comcast.net) |
| 19:10.43 | ariel_ | tzafrir_laptop, what are you trying to actually do with the asterisk box? |
| 19:10.53 | tzafrir_laptop | and will simple SIP phones know to register as "guest" before dialing to a certail number@hostname ? |
| 19:11.10 | tzafrir_laptop | ariel_, I'm trying to allow people to call me via SIP or via IAX |
| 19:11.20 | tomas- | is there like any pbx kits that come with all the necessary hardware and software and maybe phones too, etc... so a newbie to telephony can get a pbx up easily and quickly and then focus on the details later? or am i more or less left to scrounge around the internet and take a month to learn all of it? |
| 19:11.32 | ariel_ | tzafrir_laptop, how are you connected to the asterisk box? |
| 19:11.41 | Qwell | tomas-: I'm sure a consultant could gather everything for you. |
| 19:12.07 | ariel_ | tomas-, look at the dev kit and then get the iso from asterisk@home put it into a small pc and off you go. |
| 19:12.07 | tzafrir_laptop | ariel_, s/me/my asterisk/ |
| 19:12.17 | `Sauron | tzafrir: You don't have to do anything but set up extensions in the [default] context in extensions.conf |
| 19:12.18 | tzafrir_laptop | ariel_, I'm trying to allow people to call my astisk box via SIP or via IAX |
| 19:12.41 | Weezey | i think digium sells a kit |
| 19:12.44 | ariel_ | tzafrir_laptop, they can now. what is going to pickup the device on the box? |
| 19:12.50 | tzafrir_laptop | `Sauron, and if I want a context other than 'default'? |
| 19:12.56 | Katty | ariel_: haven't had a chance to get to it yet |
| 19:12.56 | `Sauron | you can't |
| 19:12.59 | tomas- | Qwell: consultant. right. |
| 19:13.14 | julianjm | tzafrir_laptop: use context=from-sip in the [general] section of sip.conf, and then define the s extension in context [from-sip] in extensions.conf |
| 19:13.15 | Katty | ariel_: between logo design and redoing website...clients...etc i've been uberbusy |
| 19:13.18 | `Sauron | incoming sip/iax calls that didn't come in through a peer, can't have a context on them |
| 19:13.20 | Uajal | Into my outgoing folder I place file beginning with: Channel: IAX2/16465157654@gwiaxt01.voicepulse.com Result: Call rejected by 66.234.228.160: no authority found. Any suggestion? |
| 19:13.29 | `Sauron | or set up a default context in extensions.conf |
| 19:13.34 | `Sauron | like julianjm said |
| 19:13.45 | tzafrir_laptop | iax does not have a general context= directive |
| 19:13.52 | Qwell | tomas-: or maybe if you explain what you want to do, somebody might be able to help you in there, give some recommended hardware, etc |
| 19:14.00 | Qwell | s/there/here/ |
| 19:14.01 | tzafrir_laptop | (I already abuse sip.conf's context directive in Rapid) |
| 19:14.14 | `Sauron | tzafrir: that's why you put the extensions in [default] |
| 19:14.15 | `Sauron | Sigh. |
| 19:14.34 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 19:14.37 | tomas- | ariel_: when you say "dev kit", youre referring to the tdm card, right |
| 19:14.46 | Qwell | tomas-: yes |
| 19:14.54 | Qwell | tdm11b, to be precise |
| 19:14.58 | *** part/#asterisk Alexis (~alexis@www.trim.it) |
| 19:15.09 | `Sauron | If you want to drop incoming calls into a specific context, just change the goto in my example |
| 19:15.13 | `Sauron | [default] |
| 19:15.14 | `Sauron | exten => dominic,1,Goto(from-sip|s|1) |
| 19:15.14 | ariel_ | tomas-, if you want to start you need to learn about asterisk or get a consultant to help you out. |
| 19:15.22 | tzafrir_laptop | `Sauron, I may not want it to call everything in my [default] . Now am I the first one with such an obvious wish? |
| 19:15.23 | *** join/#asterisk jmacz (~jmacz@63.245.86.116) |
| 19:15.35 | `Sauron | Apparently. |
| 19:16.11 | tzafrir_laptop | For instance, I don't like outsiders beling able to listen to our local voicemail |
| 19:16.52 | tomas- | is there a resource that i can get definitions for all these telephony acronyms? |
| 19:16.58 | tzafrir_laptop | They might actually find out I have no new messages!!! The horrors!!! |
| 19:17.01 | ariel_ | ~docs |
| 19:17.02 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 19:17.37 | `Sauron | tzafrir: You could drop all your other "default" calls into a specific context, and leave [default] empty other than the calls-from-the-internet thing |
| 19:17.39 | tzafrir_laptop | ~google wikipedia |
| 19:17.44 | ariel_ | tzafrir_laptop, it's alittle harder to do that you have to dial the voicemail exten you setup and then put in the box number and password. |
| 19:17.59 | ^Fenris | Failed to authenticate on INVITE to '"ksu1" |
| 19:18.10 | `Sauron | Fenris: apply the latest BV patch |
| 19:18.11 | ^Fenris | looks like a bad passwd but it used to work |
| 19:18.14 | `Sauron | or update t- cvs-head |
| 19:18.21 | `Sauron | s/t-/to |
| 19:18.27 | ^Fenris | 'Sauron: oh, thax |
| 19:18.31 | ^Fenris | thanks even |
| 19:18.49 | sivana | anyone with telco info, is RDNIS an * thing or industry thing? |
| 19:20.01 | coppice | RDNIS is an industry term |
| 19:20.17 | sivana | ok |
| 19:20.32 | sivana | my carrier was unsure about it... they're going to talk to their people |
| 19:20.34 | Darwin35 | ok anyone have a dialtone.wav |
| 19:20.57 | sivana | and I wanted to be sure I knew what I was saying :) |
| 19:21.05 | tomas- | is there such thing as a pbx card that can handle both a T1 line AND a normal POTS line? i was thinking of getting a card soon so i can fiddle around with it so i have time to get a hang of it before my T1 line gets put in |
| 19:21.14 | `Sauron | nope |
| 19:21.31 | sivana | X100P and a T100P :) |
| 19:21.43 | *** join/#asterisk Milligan (~noc@wkstn2.gnwd-noc.valuelinx.net) |
| 19:22.17 | Darwin35 | damn fricking debian box |
| 19:22.45 | Darwin35 | the hd just went south on me |
| 19:23.08 | Milligan | That sucks big time... |
| 19:23.30 | Darwin35 | it was a 1 gig and had my * setup on it |
| 19:23.40 | Darwin35 | for the mini boxe I am working on |
| 19:23.46 | Milligan | oh man..... I hope you had backups... |
| 19:23.54 | tomas- | sivana: what brand is that? |
| 19:24.00 | Darwin35 | it was a laptop drive with a 44 to 40 pin connector on it |
| 19:24.09 | Darwin35 | yeah I have a tar |
| 19:24.15 | Milligan | That is good... |
| 19:24.32 | sivana | tomas-: Diigum |
| 19:24.35 | sivana | Digium |
| 19:24.39 | hardwire | I am just going to reroute all non callerids to Echo() |
| 19:24.41 | hardwire | hows that sound? |
| 19:24.41 | Darwin35 | I make a new tar anytime I change something |
| 19:25.00 | tomas- | sivana: they look discontinued |
| 19:25.03 | Darwin35 | the tar gets shoved to a small server |
| 19:25.12 | Milligan | Hey does anyone know why when I press the messages button on a Cisco 7960 phone it rings 4 times that extension instead of going directly to voicemail? |
| 19:25.32 | Milligan | Darwin35: That is good.... |
| 19:25.34 | Darwin35 | did you set it up right |
| 19:25.42 | Milligan | Darwin35: Most people do not do that.... |
| 19:25.51 | Darwin35 | did you set the button to go right to the vm exten |
| 19:26.11 | Darwin35 | for a production drive I have to |
| 19:26.16 | Milligan | Darwin35: I believe so.... I set the message_uri to the extension number... |
| 19:26.20 | Darwin35 | I cant afford to loose the data |
| 19:26.27 | *** join/#asterisk km- (~pgrace@67.105.178.130) |
| 19:26.46 | Darwin35 | and now I have to find a drive to put it on |
| 19:26.50 | CleanerX | can anybody tell me why there issn't an CHANUNAVAILABLE sound out there |
| 19:26.59 | km- | howdy crew |
| 19:27.03 | Milligan | Darwin35: It did not always do this, I hired someone to help me work out some problems on the voicemail sertup and it has done it ever since |
| 19:27.48 | Darwin35 | ok well if you set the button on the phone to point to the exten of the vm it goes right to it |
| 19:27.56 | km- | hey guys, is there a way to reload extensions config without interrupting current calls? |
| 19:28.30 | *** join/#asterisk MisterLOVE (~MisterGoo@203.115.185.216) |
| 19:28.34 | MisterLOVE | hi people |
| 19:28.36 | Qwell | it won't reload until no calls are active, iirc |
| 19:28.39 | km- | I got in the habit of using "restart when convenient", but what's convenient for the phone system is seldom convenient for me |
| 19:28.55 | MisterLOVE | do i need a sound card to play asterisk sounds??? |
| 19:29.04 | *** join/#asterisk __cmc__ (~cmc3c@209.101.178.227) |
| 19:29.24 | tomas- | what different manufacturers make pbx cards for asterisk other than Digium? |
| 19:29.53 | Qwell | MisterLOVE: shouldn't, no |
| 19:30.07 | Qwell | tomas-: I'll be biased here. Digium cards are by far, the most supported. |
| 19:30.13 | Darwin35 | not many |
| 19:30.20 | Milligan | Darwin35: Right, but originally I had a different extension for the voicemail and he had changed it to where it is the same number as the extension of the phone, is that was is causing it? |
| 19:30.25 | Qwell | tomas-: There are other things though, that aren't "cards". Like an ATA |
| 19:30.40 | MisterLOVE | ok |
| 19:30.48 | MisterLOVE | so basically ill just install asterisk sound right? |
| 19:30.59 | MisterLOVE | because when someone dials my extensions ringing is not heard |
| 19:31.05 | Darwin35 | hmmm weird |
| 19:31.07 | Darwin35 | br |
| 19:31.11 | Qwell | asterisk-sounds is just a collection of IVR samples |
| 19:31.12 | Milligan | Darwin35: I am trying to setup this system to where I can present it as a viable product to the general public that they will be able to use.... |
| 19:31.16 | Darwin35 | brb pee break |
| 19:31.17 | MisterLOVE | ok.. |
| 19:31.22 | Milligan | k |
| 19:31.44 | Darwin35 | hay your trying to steel my busniess |
| 19:31.45 | MisterLOVE | When someone dials my extensions he doesnt hear a ring tone |
| 19:31.52 | MisterLOVE | whats wrong with that? |
| 19:31.52 | Darwin35 | I make pbx boxes also |
| 19:33.32 | Milligan | Darwin35: HEHE |
| 19:33.56 | Milligan | MisterLOVE: Mine started doing thtat too.... |
| 19:34.18 | Milligan | MisterLOVE: I can not figure it out.... |
| 19:34.35 | MisterLOVE | oic |
| 19:34.35 | MisterLOVE | really |
| 19:35.29 | Milligan | MisterLOVE: It started doing that after we made some changes to the extensions for the voice mail... It is driving me crazy... |
| 19:35.35 | MisterLOVE | really.. |
| 19:35.45 | MisterLOVE | can anyone help us with that then |
| 19:35.48 | MisterLOVE | thats my problem |
| 19:36.04 | MisterLOVE | because when someone trys to dial my extension he wont hear any ring tone |
| 19:36.06 | PTG123 | hey |
| 19:36.14 | PTG123 | whats the default password for the polycomm ip500 setup? |
| 19:36.21 | Dandan | 456? |
| 19:36.31 | *** join/#asterisk lidl (~little@81-208-60-192.fastres.net) |
| 19:36.35 | Dandan | PTG123: 456 ? |
| 19:36.37 | Milligan | MisterLOVE: It has to be there... I can not see that the changes to the voicemail causing it.... |
| 19:36.38 | PTG123 | nope |
| 19:36.47 | Dandan | PTG123: SpIP ? |
| 19:36.50 | Dandan | *SpIp? |
| 19:37.05 | Dandan | u mean website/Menu on the phone? |
| 19:38.06 | *** join/#asterisk sudhir492 (~sudhir@4.7.58.171) |
| 19:38.20 | sudhir492 | anyone using PAP2-NA here? |
| 19:38.38 | ariel_ | MisterLOVE, was this asterisk system up and running before? |
| 19:38.45 | MisterLOVE | yeah |
| 19:38.54 | MisterLOVE | i just moved the configuration to another box |
| 19:39.04 | lesouvage | ariel: thanks for the info. I live in Holland, can I buy a sipura 3000 from the USA from E-bay or do I need a ATA for the European market? |
| 19:39.13 | Dandan | did you move the hardware producing the ring tone? :D |
| 19:39.28 | sudhir492 | I guess that is same as Sipura 2000, works great for me except for one problem |
| 19:40.06 | *** join/#asterisk tomas- (~tomas@pcp0010083068pcs.frnklv01.nj.comcast.net) |
| 19:40.09 | sudhir492 | Whenever I call both the lines at the same time, I see on the asterisk window |
| 19:40.10 | ariel_ | MisterLOVE, so you installed asterisk in the other box and got all it's devices up then moved the conf files? |
| 19:40.11 | sudhir492 | Got SIP response 488 "Not Acceptable Here" back from 203.197.53.81 |
| 19:40.22 | ariel_ | lesouvage, don't know about holland. |
| 19:40.35 | *** join/#asterisk three55ml (~none@cpe-66-68-110-140.austin.res.rr.com) |
| 19:40.52 | ariel_ | lesouvage, I know that voipsupply carries them and soon will have a w/h in the EU. |
| 19:41.00 | MisterLOVE | yeah |
| 19:41.02 | MisterLOVE | thats what i do |
| 19:41.05 | MisterLOVE | i did |
| 19:41.11 | *** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net) |
| 19:41.44 | three55ml | I'm can't get my DVG-1120 to correctly send caller ID to the phone. Watching the debug on it, I can see it correctly receives it. What should I play with? Gain levels? Any other ideas? |
| 19:41.50 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net) |
| 19:42.03 | *** join/#asterisk jeffik (~jeffik@user-v7kajqe.dialup.mindspring.com) |
| 19:42.15 | *** join/#asterisk ws9455 (~ws9455@adsl-68-94-10-246.dsl.rcsntx.swbell.net) |
| 19:42.20 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 19:42.27 | MisterLOVE | i ony have 2 snome phones connected to galaxy voice |
| 19:42.33 | MisterLOVE | via my asterisk server |
| 19:42.56 | tomas- | is there anything out there that discusses how one would go about sharing a T1 line for both data (LAN) AND voice (asterisk) |
| 19:43.28 | PTG123 | anyone know the default password on the polycom? |
| 19:44.17 | ariel_ | *SpIp |
| 19:44.37 | PTG123 | so i convert that to numbers? |
| 19:44.43 | Dandan | 456 |
| 19:44.47 | Dandan | SpIp |
| 19:45.29 | Dandan | tomas-: integrated t? |
| 19:45.30 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
| 19:46.21 | SPoon_TSX | Hello there, just wondering how good Asterisk can run on Analog Phone line? How is the Voice Quality over the Analog line? Thanks you very muc. |
| 19:46.51 | ariel_ | SPoon_TSX, just fine |
| 19:46.56 | eKo1 | SPoon_TSX: eh, that has little to do with * and more to do with the equipment you use. |
| 19:46.58 | PTG123 | anhyone know how to reset a polycom? |
| 19:47.39 | SPoon_TSX | I heard people having some trouble to getting rip of the echo on the Analog connection, anyone experience the same situation before? |
| 19:48.09 | jakepdev | Spoon - it has to do with equipment |
| 19:48.25 | Dandan | PTG123: polycom.com/common/pw_item_show_doc/1,1276,2545,00.pdf |
| 19:48.29 | ariel_ | SPoon_TSX, yes there has been trouble but all in all I have better sound from analog phones then with sip. But it's all on how you set it up and which equipment you use. |
| 19:48.37 | Dandan | PTG123: http://polycom.com/common/pw_cmp_updateDocKeywords/0,1687,2545,00.pdf |
| 19:48.50 | AgiNamu | Hey, what's the shortest phone number? |
| 19:48.53 | AgiNamu | Inc. country code? |
| 19:49.08 | AgiNamu | I'm writing some code to determine if the number dialed can be a number |
| 19:49.09 | CosmicRay | 1411? |
| 19:49.16 | Darwin35 | ok queus fixed |
| 19:49.17 | CosmicRay | 10? |
| 19:49.21 | Darwin35 | voicemail fixed |
| 19:49.27 | Dandan | 1+3+7... 11 numbers |
| 19:49.31 | PTG123 | dandan doe sthat tell you how to reset though? |
| 19:49.33 | CosmicRay | though one wouldn't normally dial it that way |
| 19:49.37 | AgiNamu | Dandan, that's USA |
| 19:49.40 | AgiNamu | Globally |
| 19:49.41 | CosmicRay | one would normally dial 411 or 0 or 00 |
| 19:49.45 | SPoon_TSX | So I can use the Analog phone line for my PSTN connection and have my ASterisk Box setup as the PBX and make an VoIP call over the ethernet or Internet? Am I correct? |
| 19:49.52 | *** join/#asterisk sangee (ravi@209.250.129.135) |
| 19:49.54 | Dandan | PTG123: Yup, administrator's guide |
| 19:50.01 | jakepdev | spoon - yes |
| 19:50.03 | Milligan | Darwin35: Did you say that you setup PBX boxes? |
| 19:50.06 | Dandan | AgiNamu: it is the shortest USA has one digitcountry code... |
| 19:50.14 | AgiNamu | Dandan, so does russia |
| 19:50.19 | AgiNamu | i think |
| 19:50.19 | Dandan | AgiNamu: true |
| 19:50.23 | jakepdev | spoon - you'll need an analog interface though |
| 19:50.25 | Dandan | (7 i think) |
| 19:50.28 | AgiNamu | right |
| 19:50.30 | Qwell | but the US has 10 digit dialing. I'm sure some have less |
| 19:50.34 | AgiNamu | yea |
| 19:50.36 | Qwell | Which is what he's getting at, I'm sure |
| 19:50.42 | AgiNamu | Im wondering what the overall smallest length is |
| 19:50.47 | Dandan | Qwell: they used to.... |
| 19:50.48 | sangee | i got this message anyone know why "han_sip.c:7399 handle_request: Unable to |
| 19:50.49 | sangee | create/find channel |
| 19:50.50 | AgiNamu | Think I'll be safe with ..... 57? |
| 19:50.52 | AgiNamu | i mean, 7 |
| 19:50.53 | eKo1 | We used to have 6 digit dialing here. |
| 19:50.55 | Dandan | i think that all converted to 10 digs dialing |
| 19:51.08 | AgiNamu | Guatemala has 10 digit. But they have a 3 digit area code. |
| 19:51.10 | ariel_ | Qwell, not all us uses 10 digit lots of places still have 7 |
| 19:51.14 | sangee | i got this message anyone know why "han_sip.c:7399 handle_request: Unable to create/find channel" |
| 19:51.17 | AgiNamu | I guess I have to pop out the country code |
| 19:51.19 | Qwell | ariel_: for in the same areacode, sure |
| 19:51.21 | PTG123 | hah |
| 19:51.25 | AgiNamu | and then check the remaining number length. |
| 19:51.26 | PTG123 | says i need admin password to do a reset |
| 19:51.32 | Qwell | PTG123: nice |
| 19:51.34 | Dandan | sangee: no need to repeat |
| 19:51.37 | Qwell | PTG123: Got a large rock? |
| 19:51.41 | sangee | sorry |
| 19:51.43 | Dandan | PTG123: 456 :) |
| 19:51.44 | AgiNamu | so it's pretty sure that there's no country that's gonna allow a 3 digit dialing? |
| 19:51.46 | eKo1 | The problem is that some countries have variable length numbers. |
| 19:51.47 | AgiNamu | or 5? |
| 19:51.50 | AgiNamu | yea |
| 19:51.53 | Qwell | AgiNamu: never know :p |
| 19:51.57 | CosmicRay | AgiNamu: 411 is a valid number in the us |
| 19:52.00 | AgiNamu | I mean, in Guatemala, we have these special "17xx" numbers |
| 19:52.02 | Dandan | PTG123: http://www.voip-info.org/wiki-Polycom+Phones |
| 19:52.11 | AgiNamu | CosmicRAy, but can you dial 1411 and internationally complete the call? |
| 19:52.15 | Dandan | CosmicRay: right, but unavail from the outside |
| 19:52.18 | eKo1 | Like, I say a German number with 5 digits which was not 'special'. |
| 19:52.20 | CosmicRay | AgiNamu: probably not |
| 19:52.25 | eKo1 | s/say/saw |
| 19:52.27 | *** join/#asterisk zno (~chatzilla@ip-160-79-174-102.autorev.intellispace.net) |
| 19:52.35 | AgiNamu | hmmph. |
| 19:52.40 | three55ml | AgiNamu: Check out http://regexplib.com/DisplayPatterns.aspx?cattabindex=6&categoryId=7 |
| 19:52.57 | three55ml | AgiNamu: Not sure if you're using regexps or not, but it'd make the whole thing easier and that'd be a good starting point |
| 19:53.16 | tomas- | Dandan: from what the network tech described it as, is that there's some sort of router device that takes the whole t1 and can split between data and voice |
| 19:53.22 | tomas- | other than that, i dont know |
| 19:53.27 | PTG123 | yah that page doesn't show how to do a password reset |
| 19:53.37 | AgiNamu | three55ml, howso? |
| 19:53.41 | AgiNamu | I get a string of digits |
| 19:53.43 | eKo1 | CC is up to 3 digits and the rest can be up to 15 digits according to e.164. |
| 19:53.46 | ariel_ | tomas-, yes there are units like that |
| 19:54.01 | AgiNamu | right, here's the deal -- there's extension dialling as well |
| 19:54.02 | tomas- | ariel_: what are they considered/called? |
| 19:54.04 | AgiNamu | 5 digit codes |
| 19:54.05 | three55ml | AgiNamu: Because you can validae a string with regexps in 1 line of code |
| 19:54.08 | Dandan | tomas-: Chyoiceone? |
| 19:54.09 | ariel_ | tomas-, in fact most telco will deliver you that box as part of the package you get. |
| 19:54.11 | Dandan | *Choiceone? |
| 19:54.19 | AgiNamu | three55ml, I dont see anything there that helps. |
| 19:54.20 | Dandan | tomas-: many adtran units can do that... |
| 19:54.46 | Dandan | tomas-: ask your tech what outputs are supported? rj45/t-1/any other? |
| 19:55.06 | AgiNamu | I like regex, but regex cannot pop out a country code out of a number |
| 19:55.15 | three55ml | AgiNamu: You can setup multiple regexp rules and see if it matches them. There's just some samples there of different countries, just said it's a starting point. |
| 19:55.17 | AgiNamu | well, not a reasonable size regex |
| 19:55.18 | eKo1 | AgiNamu: Yes it can. |
| 19:55.28 | AgiNamu | three55ml, yea, but those examples are wrong and incorrect :( |
| 19:55.31 | three55ml | AgiNamu: Yes it can, very easily. |
| 19:55.34 | AgiNamu | really? |
| 19:55.35 | three55ml | AgiNamu: So write one. |
| 19:55.40 | eKo1 | You just need to compare it to list of country codes. |
| 19:55.55 | Dandan | tomas-: T-1 consists of 24 chans, when you split, you will have 23 usable (one is signaling), and you can split them to your liking - 1 data channel is 64 kbps, one voice = one line :), (gov taxes apply only to the whole t, not separate channels :) |
| 19:56.00 | AgiNamu | Yea, compare against the list |
| 19:56.00 | zno | speaking of country codes, is there any reference available for matching interntional numbers? |
| 19:56.12 | AgiNamu | three55ml, the issue I'm getting at is how do I know the smallest possible number |
| 19:56.23 | AgiNamu | To even start to create that list |
| 19:56.28 | AgiNamu | Well, I have a 50 line piece of code |
| 19:56.32 | AgiNamu | thats what gets the regex out |
| 19:56.35 | tomas- | Dandan: so i should ask for the manufacturer/model to get an idea as to what it supports? |
| 19:56.36 | AgiNamu | err, ccode out |
| 19:56.37 | eKo1 | AgiNamu: The smallest number would probably be 2 digits long. |
| 19:56.44 | AgiNamu | 71? or 11? |
| 19:56.49 | Dandan | AgiNamu: anything that has more digits than your largest extension will do? |
| 19:57.08 | AgiNamu | Well, if there can be a 5 digit number... then no. My exts. are 5 digits |
| 19:57.13 | Dandan | tomas-: more or less, ask for a salesguy who has a lot of technical knowledge... |
| 19:57.17 | eKo1 | AgiNamu: Well, I'm concluding that from e.164. |
| 19:57.31 | Dandan | AgiNamu: no, i do not think you will find a smaller numer than 5 digs... |
| 19:57.36 | AgiNamu | eko1, yea, I'm just wondering practically. e164 says up to 3 cc and 15 number |
| 19:58.02 | three55ml | AgiNamu: Are you just trying to check whether it's a valid number or extract the info from the digits? |
| 19:58.13 | eKo1 | Yes, so you can have a 1 digit cc (e.g. US) and a 1 digit number thereafter. |
| 19:58.25 | AgiNamu | three55ml, valid number. But a regex can't know that reliably |
| 19:58.34 | AgiNamu | for instance, is "13057" a valid number? |
| 19:58.40 | AgiNamu | e.164 says yes |
| 19:58.45 | Dandan | eKo1: and whom would you get? the president? |
| 19:58.48 | lordcian | has anyone successfully installed amportal? i just got it running (presumably, the messages are good) but i don't know how to load it in the brower.... |
| 19:58.59 | AgiNamu | but we know that's not actually true. is "5021777" valid? e1.64 says so, and depending on the system, it might work. |
| 19:59.01 | Goshen | installing 1.0.7RC lets see how it goes |
| 19:59.02 | eKo1 | Dandan: Well, that depends on what the phone company does with it. |
| 19:59.07 | AgiNamu | You know? screw this. I'm gonna require that extension dialling is prefixed with a * |
| 19:59.11 | AgiNamu | or # |
| 19:59.23 | Qwell | AgiNamu: there you go |
| 19:59.25 | AgiNamu | :D |
| 19:59.26 | Dandan | AgiNamu: go through your billing |
| 19:59.34 | Dandan | AgiNamu: i did that and checked :) |
| 19:59.40 | AgiNamu | Yea, I didnt check for this at first, and got all sorts of calls to Maldives, Cuba, etc. etc. |
| 19:59.51 | AgiNamu | At first I was happy cause I thought people were really into our service. |
| 19:59.57 | AgiNamu | Then I realized it was just fux0ring up :) |
| 19:59.58 | Qwell | AgiNamu: people calling Cuba, on accident, trying to dial an extension? |
| 20:00.06 | AgiNamu | or billing incoming calls incorrectly |
| 20:00.09 | eKo1 | Calling Cuba is mad expenisve. |
| 20:00.13 | AgiNamu | because our provider doesnt send DIDs with a 1 |
| 20:00.31 | AgiNamu | so someone with a 530 area code number |
| 20:00.37 | *** join/#asterisk atmel (~vlad@ruxi.dynamic.ucsd.edu) |
| 20:00.38 | AgiNamu | was getting billed for a cuba DID :S |
| 20:00.40 | eKo1 | All I know is, if the number starts with a 1, it's a US number. |
| 20:00.45 | AgiNamu | eko1 nope |
| 20:01.01 | eKo1 | well, NANP then. |
| 20:01.04 | Dandan | don't forget canada! |
| 20:01.10 | eKo1 | At least, that's my dialplan. |
| 20:01.13 | Qwell | canada is 1? |
| 20:01.14 | tomas- | Dandan: he gave me a model number "840", but i wouldnt know what kind of product it is, adtran's site has got a huge list of different products |
| 20:01.16 | AgiNamu | 1-246 Barbados |
| 20:01.27 | km- | really! |
| 20:01.41 | km- | do you have the number of a bar that might be hiring? I've always wanted to work at a tropical swim-up bar |
| 20:01.41 | eKo1 | Yeah, it's a NANP number. |
| 20:01.51 | AgiNamu | 1758785 <-- St. Lucia Cell |
| 20:01.51 | Dandan | Qwell: nobody knows, they still don't have phones out there :D |
| 20:01.59 | Qwell | ahh |
| 20:02.12 | Dandan | they just got electricity :) |
| 20:02.15 | Qwell | surely there is at least 1 phone? |
| 20:02.27 | AgiNamu | Qwell, yea. there's one phone. |
| 20:02.28 | eKo1 | A voip phone hopefully. |
| 20:02.36 | Dandan | Qwell: wouldn't be so sure, eh? |
| 20:02.43 | AgiNamu | But quebec wont allow it to be used because it doesn't say Telephone in French. |
| 20:02.49 | Qwell | I'd assume the ministry of defe...nevermind |
| 20:03.05 | *** join/#asterisk three55ml (~none@cpe-66-68-110-140.austin.res.rr.com) |
| 20:03.12 | Dandan | Qwell has quit IRC (Connection reset by ministry of defe...nevermind) |
| 20:03.20 | AgiNamu | I felt happy for Canada the other day. They told the US to piss off with their retarded missile defense system. |
| 20:03.35 | AgiNamu | Did almost as good as job as Castro or Kim Jong Il |
| 20:03.44 | Dandan | AgiNamu: lol |
| 20:03.49 | AgiNamu | Not as colourful |
| 20:03.51 | *** part/#asterisk km- (~pgrace@67.105.178.130) |
| 20:04.13 | AgiNamu | Castro put a tax on the dollar... right before it started to fall. And Kim Jong Il regularly calls Bush an "imbecile" "Saddam of the west" and other cute sayings |
| 20:04.27 | tomas- | Dandan: what i can gather, it's an AdTran Total Access device, but he said 840, which i can only find an 850 |
| 20:04.50 | Dandan | tomas-: what are you trying to find out? |
| 20:04.57 | Dandan | is it achieveable? yes |
| 20:05.35 | AgiNamu | noCharge |= !Char.IsDigit(cdr.DestinationNumber[0]); |
| 20:05.38 | AgiNamu | There, that was easy! |
| 20:05.49 | tomas- | Dandan: i just want to know what the hardware is and how it all gets set up, my exposure to communications isnt much past LAN administration |
| 20:05.52 | AgiNamu | and about 1000 times faster than regex |
| 20:06.16 | Dandan | tomas-: you are where I am... getting into telecommunication world :) |
| 20:06.22 | Dandan | being a net admin |
| 20:06.46 | coppice | get out while you still have the chance :-) |
| 20:06.55 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 20:06.56 | Dandan | coppice: i do not think so... |
| 20:07.13 | Dandan | i was kindly ASKED to replace my comapny's old phone system |
| 20:07.15 | Dandan | with voip... |
| 20:07.22 | coppice | you mean its already too late. another lost soul |
| 20:07.25 | Dandan | 70 phones, 18 outside copper lines... |
| 20:07.26 | AgiNamu | Hey, how , from the DialPlan, can you change the DestinationNumber that will be saved in the CDR? |
| 20:07.43 | AgiNamu | Say, a user Dials *69 |
| 20:07.44 | Dandan | coppice: yup... :/ |
| 20:07.45 | eKo1 | you mean dst? |
| 20:07.49 | AgiNamu | yea |
| 20:07.58 | AgiNamu | so I want to replace *69 with the actual number |
| 20:08.07 | eKo1 | Eh, you can't. |
| 20:08.15 | AgiNamu | fuck.... yet another hack I must do. |
| 20:08.29 | eKo1 | Yeah, you have to fiddle with app_dial.c |
| 20:08.38 | AgiNamu | I dont mind setting it manually |
| 20:08.45 | *** part/#asterisk j_berry (~jberry@acs-24-154-150-174.zoominternet.net) |
| 20:08.45 | AgiNamu | I'll make a SetCDRDestination app |
| 20:08.52 | spackle | Does SIP have a provision for paging? I see the Sipura phone has some config fields that seem to deal with paging. |
| 20:09.01 | AgiNamu | or... I could Dial(LOCAL) |
| 20:09.18 | eKo1 | What's wrong with having the *69? |
| 20:09.18 | AgiNamu | Anyone else thinking that perhaps Embedded Python might have been better for the extensions? :D |
| 20:09.24 | AgiNamu | eko1, if that goes to the CDR |
| 20:09.31 | AgiNamu | then I dont know where they dialed. |
| 20:09.48 | eKo1 | Well, what the heck does *69 do? |
| 20:10.06 | AgiNamu | Redials the last number dialed. |
| 20:10.09 | AgiNamu | for instance |
| 20:10.12 | zno | eKo1: It gets your gf in the position |
| 20:10.14 | Dandan | AgiNamu: *67 ? |
| 20:10.14 | eKo1 | Oh, like in the US. |
| 20:10.17 | AgiNamu | Or maybe *11 dials speedial |
| 20:10.24 | AgiNamu | I'm just making it up |
| 20:10.31 | AgiNamu | Like, maybe I allow local extensions |
| 20:10.34 | Dandan | rgr |
| 20:10.37 | AgiNamu | so they dial *11 and it calls some predefined number |
| 20:10.37 | topping | *69 calls back the last person to call you |
| 20:10.38 | spackle | There is a hack for *69 on the list or on the wiki. Needs DBput for each extension |
| 20:10.44 | AgiNamu | topping yea, that's it. |
| 20:10.52 | Dandan | topping: *67 does... |
| 20:10.53 | AgiNamu | no, i can make it work just fine |
| 20:10.59 | topping | *67 blocks caller id on the digits you are about to dial |
| 20:11.04 | Dandan | u sure? |
| 20:11.06 | AgiNamu | I just want CDR Destination to show "1305....." instead of "*69" |
| 20:11.09 | topping | ("cloaking") |
| 20:11.15 | eKo1 | Well, you still have to have a Dial(NUM) in order for it to work. |
| 20:11.23 | AgiNamu | yea, but that doesn't set the destination |
| 20:11.25 | AgiNamu | just the channel |
| 20:11.34 | topping | Dandan: was the last time i used it :) |
| 20:11.38 | AgiNamu | and channel is opaque to me. I discard it. processing results based on the channel is fragile. |
| 20:11.45 | Dandan | topping: lol :) |
| 20:12.16 | topping | that's standard PSTN mappings for those two anyway |
| 20:12.17 | eKo1 | AgiNamu: Maybe it would be a better idea to make an app for those special numbers? |
| 20:12.31 | AgiNamu | *Maybe* |
| 20:12.46 | AgiNamu | but something like *67/*69 is trivial |
| 20:12.51 | AgiNamu | to do with the dialplan |
| 20:13.33 | AgiNamu | What we need is extensible CDRs |
| 20:13.36 | AgiNamu | like... xml :) |
| 20:13.46 | tomas- | might there be a list somewhere of companies that provide asterisk-pbx based services (ie by state)? |
| 20:13.47 | AgiNamu | and just have an "AddCDRField(name,value)" app |
| 20:13.48 | eKo1 | xml? |
| 20:13.50 | AgiNamu | sure |
| 20:13.52 | topping | *69 had an interesting social effect in the states when it first came out, phone harassment went down very rapidly |
| 20:13.55 | AgiNamu | that way the parsing on the other end is nice and easy |
| 20:14.24 | AgiNamu | with XML, I dont worry if there's an extra field |
| 20:14.37 | AgiNamu | with CSV or other, an extra field causes the parsing code to crash |
| 20:14.39 | Weezey | what's the best way to make a redundant asterisk configuration? |
| 20:14.41 | Dandan | tomas-: I met with one of them... 120 dollars +.... |
| 20:14.50 | eKo1 | AgiNamu: No it doesn't. |
| 20:14.56 | Dandan | and the same knowledge you can get by calling salesppl... |
| 20:14.59 | Qwell | tomas-: I imagine theres something like that on the wiki |
| 20:15.02 | AgiNamu | eK01, well, assuming you get the field locations the same |
| 20:15.09 | zno | the whole callerid, callerid block and extra bells and whistles smells of telcos who want to add more revenue by pitting their customers against each other |
| 20:15.18 | AgiNamu | but if all of a sudden, I prepend a field to your CSV |
| 20:15.21 | AgiNamu | *boom* |
| 20:15.30 | AgiNamu | or, if you usE MySQL then to pop the data in |
| 20:15.34 | AgiNamu | *silent failure* |
| 20:15.47 | SPoon_TSX | jake: Analog Interface you mean a PSTN Card from Digiam? |
| 20:15.50 | AgiNamu | all of a sudden, you're now billing the local CallerId instead of the called number |
| 20:15.54 | AgiNamu | and since MySQL doesnt ever throw an error |
| 20:15.59 | AgiNamu | it just silently coerces data |
| 20:16.03 | eKo1 | AgiNamu: OK. I see your point. anthm was talking about this issue not so long ago. |
| 20:16.04 | jakepdev | spoon - yes |
| 20:16.07 | AgiNamu | :) |
| 20:16.12 | AgiNamu | anyways, im gonna go eat |
| 20:16.13 | AgiNamu | cya rouynd |
| 20:16.22 | Dandan | cya |
| 20:17.01 | tomas- | what kind of expansion cart/port do i need on this adtran total access box to hook up to a digium card? and isnt there an expansion card i'd need on the digium card id need? (FXS?) |
| 20:17.19 | eKo1 | The what now? |
| 20:17.22 | SPoon_TSX | I got 4 DID on 1 Analog line, it that mean I can only make 1 phone call at a time regardless how many extension I had behind the Asterisk PBX? |
| 20:18.17 | eKo1 | SPoon_TSX: You can only make one call on the analog line. |
| 20:18.39 | SPoon_TSX | That's sux. |
| 20:18.42 | *** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net) |
| 20:18.53 | eKo1 | SPoon_TSX: You only have one line. What do you expect. |
| 20:19.09 | SPoon_TSX | eKo1: Yes, hahaaa... |
| 20:19.13 | Nugget | I expect asterisk to cure male pattern baldness and prevent bad breath. |
| 20:19.35 | eKo1 | Nugget: eh, sure. |
| 20:19.48 | Essobi | I'm fricking bored. |
| 20:19.49 | SPoon_TSX | eKo1: so only solution is to get PRI or T1 to make all my extension can call out concurrently. |
| 20:20.00 | bannerman | I screwed up and used a "@" in my password, on my nufone account. So, of course, "register => username:passwordwith@symbol@server" interprets the 2nd half of my password as the servername. Is there a way to make it use an @ symbol in the password, or do I have to wait for nufone to change my pw? |
| 20:20.02 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 20:20.06 | topping | Nugget: can you explain in great detail how it can do that? I haven't read the manual yet. |
| 20:20.11 | eKo1 | SPoon_TSX: That depends on how many extensions you're going to use. |
| 20:20.31 | Essobi | bannerman Umm.. hack chan sip to except a quote character? |
| 20:20.32 | Essobi | :) |
| 20:20.33 | eKo1 | bannerman: What for nufone. |
| 20:20.35 | Nugget | topping: it only works if you are running asterisk in freebsd, naturally. |
| 20:20.41 | topping | :-) |
| 20:21.07 | bannerman | Everyone: qq. |
| 20:21.18 | tomas- | Dandan: so where are you at with all this? do you actually have a running pbx yet? |
| 20:21.28 | Essobi | Someone donate one of those new T1 cards to me, and I'll hack * to use it on FreeBSD. |
| 20:21.50 | Essobi | they got FreeBSD drivers |
| 20:22.17 | eKo1 | Who is 'they'? |
| 20:22.24 | Essobi | lemme look it up |
| 20:23.44 | bannerman | Is there a better way to get ahold of Nufone? Already put a call and an email to fastsupport in.. no response yet. |
| 20:23.56 | SPoon_TSX | What got now is a 4 DID as roll-over number and 1 primailry number for my new office, but they all comes in Analog. Then I need 10 extension on my PBX to serve both VoIP and PSTN call. Which Digiam Card you recommend for my application? |
| 20:24.08 | Mw3 | bannerman: everybody is at von :) |
| 20:24.08 | eKo1 | bannerman: Isn't JerJer the owner of Nufone? |
| 20:24.10 | *** join/#asterisk miguel3239 (~myoung@ip66-105-121-66.z121-105-66.customer.algx.net) |
| 20:24.50 | bannerman | eKo1: might be, his name is Jeremy anyhow |
| 20:24.58 | Essobi | Sangoma A101u E1/T1 PCI i |
| 20:25.04 | Essobi | that's the card |
| 20:25.10 | Essobi | there's * support already |
| 20:25.12 | bannerman | Mw3: thanks :-P |
| 20:25.17 | Essobi | and it's got FreeBSD drivers |
| 20:25.23 | bannerman | I guess I'll just get another account. |
| 20:25.33 | Essobi | So potentially FreeBSD * T1/E1 could work.. |
| 20:25.43 | tomas- | anyone know how i might go about hooking up a pbx server to an IAD (serving both data and voice)? |
| 20:25.45 | eKo1 | potentially... |
| 20:26.04 | Essobi | eKo1 Yup.. Jer runs NuPhone |
| 20:26.34 | Essobi | eKo1 Like I said.. Donate a card to me, and I'll hax0rs it. :) |
| 20:26.51 | eKo1 | Essobi: Go to VON and steel one. |
| 20:26.57 | Essobi | LOL.. |
| 20:27.01 | Essobi | No, my boss might see me. |
| 20:27.08 | Essobi | He's there. |
| 20:27.17 | Essobi | Either way thou.. |
| 20:27.19 | Skysky | hi, can someone give me some tips on starting festival server at bootup |
| 20:27.25 | Essobi | http://www.soekris.com/bundles.htm <-- those look pretty tight. |
| 20:27.47 | Essobi | Beats the shit out of 3K for a Cisco 2600 with a PRI to loop up into a PBX. |
| 20:28.16 | topping | Essobi: can that thing run asterisk? |
| 20:28.18 | *** part/#asterisk zpn (~xpn@dhcp-152.digium.com) |
| 20:28.21 | boch | whats beats? |
| 20:28.22 | Essobi | topping Yup. |
| 20:28.26 | topping | nice |
| 20:28.40 | Essobi | 8.5 X 6 x 1.3" |
| 20:28.45 | Essobi | pretty small foot print too |
| 20:28.56 | Essobi | http://www.voip-info.org/wiki-Asterisk+hardware+Soekris |
| 20:29.00 | boch | the beats, the sucesors of the beatles |
| 20:29.17 | Essobi | Some guy on here says he's got 12 SIP phones and a T1 running on it. |
| 20:29.49 | Essobi | And it's got a hardware watchdog.. Not too shabby. |
| 20:30.02 | Dandan | ok, CVS or stable? |
| 20:30.08 | Dandan | quick quick quick |
| 20:30.17 | eKo1 | stable |
| 20:30.19 | Essobi | 1 WIC, 3 ether, |
| 20:30.19 | topping | can someone explain the difference between context and section in the configs? i understand context just fine, but sections elude me |
| 20:30.22 | Essobi | 1 usb |
| 20:30.28 | Essobi | and 2.5" IDE header. |
| 20:30.43 | Essobi | Dandan CVS |
| 20:30.51 | Dandan | ok, 1:1... |
| 20:30.56 | Dandan | any1 else? |
| 20:31.01 | eKo1 | actually, cvs or stable doesn't make sense. |
| 20:31.05 | Essobi | I love bleeding edge |
| 20:31.08 | Dandan | eKo1: y? |
| 20:31.10 | topping | Essobi: that usb is nice so you can plug in cd-rom when you need to upgrade and not have to open the lid |
| 20:31.11 | `Sauron | Does anyone know of a company that makes DSP-based T1 cards? |
| 20:31.14 | `Sauron | channelised, etc |
| 20:31.17 | eKo1 | It should be cvs stable or cvs head. |
| 20:31.26 | Essobi | topping Or a USB keychain. |
| 20:31.31 | topping | ah good point |
| 20:31.34 | Dandan | eKo1: heh |
| 20:31.53 | eKo1 | USB keychains roX0rs |
| 20:31.56 | Essobi | Sauron Be prepared to pay $$ for DSP cards. |
| 20:32.05 | `Sauron | Essobi: I don't care about $$$'s |
| 20:32.34 | `Sauron | we need to replace our digi rasfire cards |
| 20:32.39 | `Sauron | which were $6k/ea |
| 20:32.54 | eKo1 | What are those cards for? |
| 20:33.09 | Essobi | In that case.. |
| 20:33.10 | Essobi | http://www.sangoma.com/ |
| 20:33.14 | Essobi | I believe they do.. |
| 20:33.21 | Essobi | What support they have.. I don't know. |
| 20:33.27 | `Sauron | Hum. |
| 20:33.33 | `Sauron | linux a must |
| 20:33.38 | `Sauron | we'll see |
| 20:33.48 | SPoon_TSX | Just wondering is it possible to have 1 DID for income but 4 DID for out going in Analog line? |
| 20:33.57 | Essobi | It's got linux and *bsd for everything they make I believe. |
| 20:34.19 | Essobi | SPoon_TSX You can't set your caller ID on an analog line. |
| 20:34.27 | *** join/#asterisk tomas- (~tomas@pcp0010083068pcs.frnklv01.nj.comcast.net) |
| 20:34.28 | eKo1 | SPoon_TSX: 4 DID for outgoing? |
| 20:34.38 | eKo1 | That makes no sense. |
| 20:34.42 | SPoon_TSX | I am totally confuse now!! = ( |
| 20:34.47 | Essobi | Only PRIs support that.. maybe capi stuff.. I don't know. |
| 20:35.24 | `Sauron | Essobi: None of them look like they're channelised |
| 20:36.18 | SPoon_TSX | What happen is I have a new office and my boss what to use Asterisk as the PBX so that we can use VoIP. However, I am having some trouble on to understand how many Analog line I need to help at least 4 concurrent call. Any suggestion? (My boss is so chea that band my PRI idea already.) |
| 20:37.13 | Ash | DIDs have nothing to do without outbound calls |
| 20:37.27 | TheBear | anyone on the lists.digium.com admin staff, I've tried twice to subscribe to asterisk-users and no sucess ?? |
| 20:37.32 | Ash | or inbound, other than that's hwo your pbx knows what number to send the incoming call to |
| 20:38.15 | Ash | SPoon_TSX: I'd just say get 4 analog lines. |
| 20:38.26 | tomas- | what exactly does an "integrated access device" do? relating to a T1-line? |
| 20:38.35 | Essobi | Sauron http://www.sangoma.com/wanpipe.htm |
| 20:38.40 | Essobi | A101c |
| 20:39.03 | SPoon_TSX | Ash: Can I have all 4 analog on the same number? which mean can I ask my telcom company to route whatever call to 1 number and ring those 4 Analog line? |
| 20:39.34 | Ash | SPoon_TSX: sure, they will do that for you |
| 20:39.42 | `Sauron | Essobi: And I bet you they use libspan to de-channelise it for * |
| 20:39.54 | `Sauron | however, that doesn't work for faxing, etc. |
| 20:39.55 | Ash | usually it's some sort of rollover, where if line 1 is busy they send to line2, etc. |
| 20:39.56 | SPoon_TSX | Ash: If so, which card should I get from Digium? |
| 20:40.15 | Ash | SPoon_TSX: no idea about the digium cards, I don't use any analog lines (only PRIs) |
| 20:40.33 | SPoon_TSX | Ash: You are luckie! = ) |
| 20:40.38 | `Sauron | spoon: get a tdm400, with 4 fxo modules |
| 20:40.49 | `Sauron | and ask your telco to hunt the phone numbers |
| 20:40.57 | tomas- | correct me if im wrong, but do the Digium T1 cards connect directly to a T1-line, making an integrated access device (like an AdTran) unnecessary? |
| 20:41.17 | `Sauron | tomas: depends on whether the digium cards have a csu/dsu onboard or not |
| 20:41.18 | Ash | tomas-: that is correct |
| 20:41.23 | Ash | you can have a CSU if you want, I suppose |
| 20:41.29 | spackle | Does SIP have a provision for paging? I see the Sipura phone has some config fields that seem to deal with paging. |
| 20:41.37 | `Sauron | Essoi: Basically, I need a drop-in replacement for the Digi RasFire card. Grmph. |
| 20:41.56 | Ash | but I've got my te110p plugged directly into the smartjack in our demarc with no problems. |
| 20:41.59 | SPoon_TSX | Sauron: Thanks. |
| 20:42.04 | SPoon_TSX | Thank everyone! |
| 20:42.12 | Ash | CSUs come in handy if you're having line troubles and you want to throw an analyzer on the line, though |
| 20:42.16 | Essobi | Sauron No idea bub.. Wish I could help. |
| 20:42.18 | Ash | since you can do that without taking the span down |
| 20:42.47 | tomas- | so then if i get an AdTran from my service provider for my T1, i can just use an analog Digium card (I only need 4 analog lines anyway) hooked up to an expansion FXS slot on the AdTran? Is there any way to hook up the Digium T1 cards to the AdTran's? |
| 20:43.22 | Ash | why not just get a digium T1 card and forget about the adtran? |
| 20:43.32 | Goshen | can someone give me their register line from sip.conf for Broadvoice? the one on the wiki doesn't seem to be working |
| 20:43.49 | `Sauron | Goshen: follow the first link in the bv settings page on the wiki |
| 20:43.59 | `Sauron | it has a full accurate example conf |
| 20:44.22 | tomas- | Ash: well they're giving it to me anyway, and 2) i'd like to try the pbx out before i get the t1 installed and 3) how would i get data+voice out of the same t1? |
| 20:44.23 | Goshen | Sauron: that is a nice setup page...it doesn't seem to be working though, I will keep at it...perhaps they have not activated my service yet |
| 20:44.33 | `Sauron | Goshen: it works. |
| 20:44.36 | `Sauron | Trust me |
| 20:44.44 | `Sauron | especially with the invite auth problem |
| 20:45.04 | `Sauron | If you can't get it to work, cut+paste the example conf, and replace the field values |
| 20:45.11 | Ash | tomas-: you route them separately with the asterisk box, I think it's covered in voip-info's wiki somewhere |
| 20:45.23 | Ash | I haven't ever had to do that myself, but it shouldn't be any big deal |
| 20:45.43 | Ash | if the adtran has a T1 interface on it, that works too; you could just plug from there into the digium T1 card |
| 20:45.50 | Ash | then ethernet out of the adtran for data, I'd imagine? |
| 20:45.51 | tomas- | Ash: well whats wrong with just using the adtran and an analog FXO card? |
| 20:46.19 | Ash | i thought you said you wanted a few channels of data out of the T1? |
| 20:46.22 | Ash | er of voice |
| 20:46.35 | tomas- | Ash: well isnt the T1 interface hooked up to the data pipe out? |
| 20:46.42 | tomas- | Ash: yeah |
| 20:46.43 | Ash | yes, but usually it passes through |
| 20:47.03 | Ash | so basically the adtran sucks off the channels it needs and hands the rest through it to your pbx |
| 20:47.35 | Ash | so you need a T1 interface to hook to the adtran |
| 20:47.39 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 20:47.40 | `Sauron | hum |
| 20:47.47 | `Sauron | the adtran sounds like an add/drop csu |
| 20:47.57 | Ash | at least, if it's anything like the boxes I've had that have done that in the past |
| 20:48.21 | tomas- | so if the asterisk box takes the t1 in, would that also act as my network firewall? |
| 20:48.25 | TheBear | where on wiki can I find a complete list of sip.conf pairs ? |
| 20:48.36 | tomas- | if i wanted it to,that is |
| 20:48.37 | Ash | tomas-: yes, it would act as a firewall/router |
| 20:48.45 | Ash | that's what I'd do, to be honest. |
| 20:49.01 | Ash | you get a lot more features from linux routing and firewalling than you're likely to from that adtran box |
| 20:49.35 | tomas- | one adtran model has ethernet on it too |
| 20:49.36 | *** join/#asterisk john867530 (0@207.177.124.3) |
| 20:49.43 | `Sauron | Yawn. |
| 20:49.52 | tomas- | but yeah id have a linux firewall anyway |
| 20:49.58 | `Sauron | Sometimes, people's ignorance amazes me. |
| 20:50.02 | Ash | so, do it in one box and be done with it |
| 20:50.09 | SexyKen | Does anyone here know how to 'track' calls/join them together in the CDR so that we know certain 'calls' were connected to other ones? For instance, You dial 866 TESTING which enters you into an IVR. You select "1" for Test Further. Then you wait in a queue. When the queue clears, you are connected to an agent. -- That's a total of 3 calls that Asterisk logs into the CDR, but there's no way to know that the call is from the same origin |
| 20:50.13 | Ash | the CSU doesn't really buy you that much. |
| 20:50.43 | `Sauron | SExyKen: I suspect nobody does, since you've asked the question at least 3 times, and not gotten an answer yet. |
| 20:50.52 | john867530 | Anybody know if there is a manager interface to meetme & meetmeadmin>? |
| 20:50.52 | tomas- | so the digium t1 cards can basically replace the adtran thing AND provide a PBX and firewall at the same time? |
| 20:50.59 | tomas- | so the digium t1 cards can basically replace the adtran AND provide a PBX and firewall at the same time? |
| 20:51.00 | Ash | tomas-: exactly |
| 20:51.09 | Juggie | SexyKen, how many times are you going to ask that god damn question |
| 20:51.11 | `Sauron | tomas: You don't need a firewall with voice calls |
| 20:51.11 | SexyKen | `Sauron - Eventually someone will be in here that will know. |
| 20:51.19 | Juggie | if you dont like how the cdr tracks calls do something about it |
| 20:51.30 | Ash | `Sauron: he means for internet stuff |
| 20:51.32 | tomas- | `Sauron: i need it with my LAN |
| 20:52.05 | `Sauron | Yeah, but running voice channels into a linux box running *, has absolutely nothing to do with "internet stuff" |
| 20:52.23 | `Sauron | Unless the zaptel drivers have some magical add/drop capabillity. |
| 20:52.26 | Ash | `Sauron: he's also going to be handling his data channels on the T1 through the same box potentially |
| 20:52.29 | *** join/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 20:52.51 | Ash | tomas-: check on voipinfo to be sure about the data stuff. |
| 20:52.59 | `Sauron | Ash: <`Sauron> Unless the zaptel drivers have some magical add/drop capabillity. |
| 20:53.15 | *** join/#asterisk brettnem (~mive29@user-0ccsr2l.cable.mindspring.com) |
| 20:53.24 | Goshen | when trying to dial out over broadvoice I get this error... -- Got SIP response 604 "Does not exist anywhere" back from 147.135.0.128 |
| 20:53.46 | SexyKen | •Juggie• not everyone knows c like the back of their hand. |
| 20:53.47 | SexyKen | Dildo. |
| 20:53.50 | Ash | `Sauron: http://www.voip-info.org/wiki-Asterisk+Data+Configuration |
| 20:53.58 | Ash | "Zaptel PRI cards (T100P, E100P, T400P, TE405P, TE410P) can be configured for data (HDLC "family") communication (WAN router), as well as hybrid configurating where some T1/E1 timeslots will be used for data and others for voice communication. |
| 20:54.02 | Ash | " |
| 20:54.04 | tomas- | okay so other than if i have a large amount of phone lines, why would i go the route of using the t1 card instead of an analog card (assuming both suffice for my input capacity)? |
| 20:54.31 | Ash | tomas-: an analog card would give you one line |
| 20:54.34 | `Sauron | Comment: Everything works with kernels below 2.4.20, thereafter Cisco HDLC is broken in Zaptel driver. |
| 20:54.36 | `Sauron | Hehn. |
| 20:54.44 | Ash | heh heh |
| 20:54.48 | Ash | nice! |
| 20:55.05 | *** join/#asterisk darby_t (~tom@dnc187.neoplus.adsl.tpnet.pl) |
| 20:55.13 | SexyKen | Is it true that a T1 line only has a single data channel but has multiple voice channels? |
| 20:55.28 | Ash | tomas-: you can't just plug a T1 into an analog card |
| 20:55.30 | `Sauron | SexyKen: Both statements are "Wrong" |
| 20:55.32 | tomas- | Ash: i mean if i got the digium quad-port analog card hooked up to the adtran (with an FXS adapter), versus just using the digium T1 card by itself |
| 20:55.48 | SexyKen | Sauron, then what is correct? |
| 20:55.58 | `Sauron | a T1 (DS1) consists of 24 DS0 (56k) channels |
| 20:55.58 | Ash | tomas-: you're doing one less D/A conversion if you just do straight T1 |
| 20:56.05 | Essobi | Can zaptels do dynamic re-allocation of timeslots? |
| 20:56.25 | tomas- | Ash: and whats the advantage of that, paying twice as much for the card? |
| 20:56.33 | Dandan | `Sauron: 64k channels |
| 20:56.34 | Ash | better voice quality |
| 20:56.36 | `Sauron | As such, a T1 can hold either 23 voice channels (1 DS0 for signalling), or 24*56k worth of data |
| 20:56.39 | SexyKen | 24*56 = 1344 |
| 20:56.41 | Dandan | 24 * 64 = 1.544 M |
| 20:56.48 | `Sauron | HRm, right |
| 20:56.48 | tomas- | $350 vs. $600 |
| 20:56.52 | Ash | tomas-: and possible future expandability |
| 20:56.54 | `Sauron | E1 is 32 56k channels |
| 20:56.58 | LoRez | `Sauron: 64kbps per channel |
| 20:57.03 | Dandan | `Sauron: *64... |
| 20:57.04 | Ash | our card wasn't $600, tomas |
| 20:57.08 | `Sauron | yeah yeah I got it the first time |
| 20:57.09 | `Sauron | gah |
| 20:57.14 | Dandan | 32 * 64 = 2048 m |
| 20:57.17 | Ash | `Sauron: 64k |
| 20:57.19 | tomas- | Ash: what was it? |
| 20:57.19 | Ash | *g* |
| 20:57.25 | Dandan | :D |
| 20:57.26 | `Sauron | Ash: eat me |
| 20:57.27 | SexyKen | So hmmm. |
| 20:57.28 | Ash | tomas-: looking |
| 20:57.35 | tomas- | the make/model, that is |
| 20:57.35 | SexyKen | Why does a T1 cost so much? |
| 20:57.39 | `Sauron | There are some providers that don't give 64k channels |
| 20:57.46 | Dandan | SexyKen: reliability |
| 20:57.46 | `Sauron | the non-"clear channel" crap |
| 20:57.50 | puppet | sexyken: cause no one needs it anymore ;d |
| 20:57.52 | Dandan | what's the CIR on this T-1? |
| 20:57.56 | Ash | tomas-: oh nevermind, yes it was. We must have gotten it on sale. |
| 20:58.02 | Dandan | puppet: I still do |
| 20:58.08 | Dandan | puppet: many ppl does |
| 20:58.14 | Dandan | *do |
| 20:58.17 | puppet | dandan: why really when there is other solutions? |
| 20:58.21 | SexyKen | Dandan - I'm about 110 yard away from my Centrol Office, think they'll install a T1? |
| 20:58.23 | Ash | voipsupply.com is where we got it |
| 20:58.29 | tomas- | Ash: so why should i spend the extra money, assuming i dont forsee myself needing more than 4 phone lines? |
| 20:58.30 | `Sauron | sexyken: yes |
| 20:58.30 | Dandan | puppet: what other solutions? |
| 20:58.49 | SexyKen | How much would it cost to install a T1? |
| 20:58.52 | puppet | dandan: fiber/citylan or something and run all voip |
| 20:58.53 | `Sauron | depends |
| 20:58.54 | Dandan | SexyKen: 110 yards? 100 meters? ask them to peer you with an ethernet cable |
| 20:59.00 | Ash | tomas-: it's a tiny amount of money to pay now if it ever turns out you need more channels. |
| 20:59.05 | tomas- | SexyKen: i think a T1 is just over the normal telco infrastructure, you should be able to get it mostly anywhere, definitely outside of dsl areas |
| 20:59.08 | Dandan | SexyKen: usually it is waived if you sign up for 3 yr contract or so |
| 20:59.08 | `Sauron | point to point T1 just gosts the loop charge |
| 20:59.12 | Ash | because with one PRI you can go up to 23 channels of voice |
| 20:59.18 | `Sauron | then you pay extra for bandwidth with someone |
| 20:59.19 | SexyKen | Dandan, 110 yards. About a football field and the touchdowns. |
| 20:59.27 | Ash | do whatever you want, I really could care less :) |
| 20:59.30 | Dandan | puppet: fiber? citylan? when was the last time you checked prices on those? |
| 20:59.44 | puppet | dandan: it dont cost that much here in sweden |
| 20:59.45 | Dandan | SexyKen: yeah :) max ethernet segment is ~ 110 yds |
| 20:59.53 | puppet | dandan: like 0 :) |
| 20:59.59 | Dandan | puppet: like where?!? |
| 21:00.03 | `Sauron | max. ethernet is 300-ish feet |
| 21:00.08 | puppet | dandan: Sweden :) |
| 21:00.10 | `Sauron | slightly less than 100 meters |
| 21:00.13 | Dandan | `Sauron: foot being .3 m :) |
| 21:00.24 | zigman | get lwl singlemode |
| 21:00.26 | zigman | 70km |
| 21:00.28 | `Sauron | Dandan: foot being slightly less than .3 m |
| 21:00.39 | `Sauron | Don't argue metric numbers with me |
| 21:00.41 | SexyKen | Hmm. |
| 21:00.55 | Dandan | `Sauron: 328 feet to be exact... where it has been specified in M then converted to feet, btw. I saw segments 150 m long |
| 21:00.59 | Dandan | working just fine |
| 21:01.00 | tomas- | Ash: im getting a 7-liner package from my provider, tho i dont see myself using more than 4 -- 1 fax, 1 for business A, 1 for business B and my home line on the POTS |
| 21:01.14 | tomas- | Ash: besides, i have a POTS line for my home phone, id like to hook up to the pbx |
| 21:01.17 | SexyKen | So what would the total estimated cost per month be for a T1? |
| 21:01.20 | Dandan | puppet: FU :P we are talking the US now :D |
| 21:01.26 | puppet | dandan: hahaha |
| 21:01.28 | puppet | dandan: ;D |
| 21:01.35 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 21:01.37 | tomas- | SexyKen: where im getting it, its $300/mo, but for cost effectiveness it needs to be $400 |
| 21:01.37 | Dandan | i do not have a 10 mbit connection to my home for 20 dollars (Euro) a month... |
| 21:01.40 | puppet | dandan: i promise that a t1 in sweden costs more then fiber/citylan |
| 21:01.46 | bannerman | `Sauron: Regarding Point to Point T1.. I've got two facilities about 10 miles apart, but in different area codes.. best quotes I've come up with have been in the neighborhood of $1400/mo.. am I asking the wrong questions you think? |
| 21:02.02 | Dandan | puppet: cos the freakin' sweden is like korea of europe |
| 21:02.09 | Dandan | everyone has fibver/b-band |
| 21:02.09 | xkev | bannerman that's way too high |
| 21:02.11 | `Sauron | bannerman: If you have to cross LATA boundaries, you're screwed |
| 21:02.12 | puppet | dandan: ;p |
| 21:02.22 | puppet | dandan: i got 6mbit adsl at home 2mbit up |
| 21:02.30 | `Sauron | but for 10 miles, that's too high |
| 21:02.31 | tomas- | SexyKen: actually, what im getting is fractional t1, but they dont meter it |
| 21:02.32 | puppet | dandan: 4 static ips |
| 21:02.42 | Dandan | bannerman: ask for 2 t's and do vpn |
| 21:02.43 | SexyKen | 6mbit is faster than T1 isn't it? |
| 21:02.43 | `Sauron | we have T1's > 10miles for $370/month |
| 21:02.47 | `Sauron | 377/month |
| 21:02.50 | puppet | sexyken: yeah |
| 21:02.55 | puppet | sexyken: ;p |
| 21:03.06 | Dandan | cheapest t-1 here (CT) - 500 a month |
| 21:03.12 | SexyKen | Why would someone choose DSL over T1? |
| 21:03.15 | bannerman | `Sauron: I was afraid of that. One of the telco guys told me I'd have to backhaul through a nearby major city, which all told crosses 4 area codes and 3 telcos- verizon, qwest, sprint |
| 21:03.19 | `Sauron | of course we get bulk discount |
| 21:03.19 | Dandan | SexyKen: 4 times faster |
| 21:03.27 | `Sauron | we have > 75 T1's just in austin alone |
| 21:03.28 | xkev | SexyKen, because it has no reach limit |
| 21:03.44 | `Sauron | bannerman: Ugh. |
| 21:03.46 | xkev | T1 is a flavor of DSL, for the literal acronym |
| 21:03.46 | Dandan | `Sauron: there you go... |
| 21:03.54 | `Sauron | see if you can get line-of-sight, and do wireless? :) |
| 21:03.55 | tomas- | Ash: so if i want to hook up my POTS line to my pbx and im gonna get an adtran anyway, probably with the FXS expansion cards already, and wouldnt need more than 4 phone lines, wouldnt it make sense to just get the analog digium card? this is a soho environment, mind you |
| 21:04.07 | FuriousGeorge | i have a stupid question. i just compiled asterisk and it appears to have no interface, cli gui or otherwise. assuming i wanted to set this up only w/ voip services, where would i start? |
| 21:04.08 | Dandan | puppet: I have 5 mb/512 kb cable |
| 21:04.08 | bannerman | `Sauron: I wish. There's a mountain. |
| 21:04.11 | xkev | `Sauron, yeah, get a wireless ds3 set off ebay for $900 |
| 21:04.16 | puppet | dandan: :) |
| 21:04.18 | xkev | two sets then :) |
| 21:04.23 | `Sauron | hehe |
| 21:04.25 | bannerman | `Sauron: This is the most beautiful place in the world to live, and the most horrible place to try to communicate. |
| 21:04.43 | Bacon | FuriousGeorge: With a text editor in /etc/asterisk |
| 21:04.46 | CosmicRay | bannerman: huh, I thought I lived there :-) |
| 21:04.48 | CosmicRay | bannerman: where are you? |
| 21:04.48 | `Sauron | bannerman: CO or new england? |
| 21:04.53 | Dandan | FuriousGeorge: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 21:05.07 | Dandan | FuriousGeorge: http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 21:05.08 | bannerman | `Sauron: Almost. Columbia Gorge, in Washington state. |
| 21:05.10 | Ash | tomas-: if you've already got something worked out from your provider like that, then yeah; just go with analog cards. |
| 21:05.20 | bannerman | Colorado is a close 2nd. |
| 21:05.28 | tomas- | Ash: thanks for the help |
| 21:05.42 | Dandan | bannerman: do vpn over two separate t's... |
| 21:05.56 | Ash | tomas-: np! |
| 21:05.56 | Ash | :D |
| 21:06.05 | FuriousGeorge | thats what i thought. then i would run asterisk, asd see if i can log into my server using a softphone (i dont have anytrhing yet besides * itsel yet) |
| 21:06.08 | CosmicRay | I dunno, I am skeptical of ranking places in terms of beauty... I think that beauty can be expressed in different ways. though there are clearly ugly places. Gary, IN for one. |
| 21:06.10 | bannerman | Dandan: that's what I'll have to do, I guess. Right now we have satellite at the other facility, which just kills us with it's high latency. |
| 21:06.11 | `Sauron | dandan: I bet you it would be almost as expensive for him as doing it the other way |
| 21:06.20 | FuriousGeorge | dandan: thanks |
| 21:06.39 | bannerman | `Sauron: Actually, I was able to get a T1 here through a local ISP for $395, and I can get one at the other place for $365 |
| 21:06.39 | CosmicRay | though they tend to be ugly because of human activity |
| 21:06.45 | Dandan | `Sauron: hmm, it is just an alternative... |
| 21:06.45 | `Sauron | hum |
| 21:06.49 | Dandan | bannerman: g'luck :) |
| 21:06.52 | Dandan | FuriousGeorge: np |
| 21:07.10 | bannerman | But when I'm explaining that to the boss, we're talking about upgrading from $120/mo to $700/mo for Internet.. |
| 21:07.11 | puppet | Anyone more then me that have problems with that MoH is REAL messy? Not Once ive been able to get a clear stream |
| 21:07.24 | bannerman | obviously there is more to it, but looking at the numbers, it's bad. |
| 21:07.26 | Dandan | bannerman: what would you use it for? |
| 21:07.44 | bannerman | Dandan: voip, file sharing, database access |
| 21:07.53 | Dandan | bannerman: no. of clients? |
| 21:08.09 | bannerman | Very few. Like 2 phone lines max. |
| 21:08.16 | Dandan | get split t or dsl |
| 21:08.22 | bannerman | DSL.. I wish. |
| 21:08.35 | Dandan | cable? |
| 21:08.36 | bannerman | That would solve all of my problems. I could go home a happy man. |
| 21:08.42 | bannerman | I wish :-P |
| 21:08.47 | CosmicRay | cable? |
| 21:08.48 | bannerman | really out in the sticks here |
| 21:08.54 | CosmicRay | heh |
| 21:09.18 | CosmicRay | bannerman: hey, you're worse than me. dsl became available at my house a few months ago. before that, not even isdn was an option. |
| 21:09.21 | CosmicRay | pots or nothing. |
| 21:09.36 | bannerman | CosmicRay: Ouch, I thought you could get ISDN anywhere. |
| 21:09.41 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 21:09.47 | bannerman | CosmicRay: fortunately, I get comcast at my home. |
| 21:09.49 | roamer323 | puppet - what codec r u using? |
| 21:10.00 | *** join/#asterisk alfonso_ (~alfonso@166.114.250.52) |
| 21:10.03 | Nugget | isdn is metered a lot of places, so even if you can get it you'll end up paying $500 a month to actually use it. |
| 21:10.05 | CosmicRay | bannerman: sprint local sucks |
| 21:10.23 | Nugget | my mom lives out in the boonies and all she can get is that two-way satellite stuff |
| 21:10.28 | bannerman | CosmicRay: Amen. |
| 21:10.40 | CosmicRay | bannerman: I had comcast for awhile in indianapolis. was unreliable when it was still @home, and when it became comcast, they kept the unreliability and capped the speed at 1/3 of what it was before |
| 21:10.48 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 21:11.10 | CosmicRay | Nugget: that sucks too. I had fixed wireless before we got dsl, but it was unreliable, both due to insurmountable trees on our property and the distance from the tower |
| 21:11.14 | bannerman | when I first got Comcast here in Washington it was down on a regular basis, but since then they've upgraded our bandwidth twice (I think I get 3Mbps now) and their reliability has been amazing |
| 21:11.38 | alfonso_ | hi all |
| 21:11.42 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
| 21:11.46 | bannerman | my wife and I are certified MMORPG addicts, if it's down for a minute during primetime, it's a disaster. haven't had any outage in the last two months. |
| 21:11.48 | CosmicRay | bannerman: so you have had to deal with sprint local? I didn't think there were many sprint towns outside the midwest |
| 21:11.52 | CosmicRay | heh |
| 21:12.13 | heison | does anyone know what this may referr to? |
| 21:12.14 | heison | Mar 11 16:07:41 NOTICE[1159]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
| 21:12.14 | heison | cause 3 means AST_CAUSE_NO_ROUTE_DESTINATION |
| 21:12.23 | heison | but i can't figure out what has possibily gone wrong... |
| 21:12.25 | alfonso_ | need help with my new TDM400P w/4 FXO mods. |
| 21:12.37 | heison | I have replaced my 7960 with another 7960 |
| 21:12.56 | heison | it's the same IP, and exten, just different MAC |
| 21:12.59 | bannerman | CosmicRay: yep.. it's not the greatest. Fortuntely we're just using a T1 and VoIP :) |
| 21:13.07 | alfonso_ | heison I got the same error today |
| 21:13.13 | CosmicRay | bannerman: I'm impressed they managed to get that done |
| 21:13.16 | heison | comparing SIPXXXXXX.cnf between the 2 are identical |
| 21:13.22 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 21:13.26 | heison | alfonso_: what was it? |
| 21:13.43 | heison | it works for other 7960s, but just this one... |
| 21:13.45 | alfonso_ | qualify=300 parameter |
| 21:13.45 | CosmicRay | bannerman: here's my account of how they fscked up my dsl install: http://changelog.complete.org/node/143 |
| 21:14.05 | alfonso_ | i set to low |
| 21:14.22 | heison | i have qualify=yes |
| 21:14.31 | heison | what should i change it to? |
| 21:15.04 | terrapen | anyone work at nufone? |
| 21:15.07 | alfonso_ | to qualify=2000 |
| 21:15.13 | heison | terrapen: try #nufone |
| 21:15.17 | terrapen | did. |
| 21:15.23 | CosmicRay | bannerman: I remember working for a regional isp back in the days of expanding dialup service to new locations. sprint towns were always the last to get it because of issues with them. |
| 21:15.28 | Essobi | heison Sap firmware on the phones? |
| 21:15.31 | terrapen | i desperately need someone to add funds to my account |
| 21:15.31 | Essobi | Same? |
| 21:15.34 | terrapen | paypal'ed them |
| 21:15.49 | heison | Essobi: yes, same firmware |
| 21:15.52 | terrapen | this makes me angry because i've always stood up for nufone when people talk smack about them |
| 21:15.52 | heison | 7.3 |
| 21:15.54 | Essobi | Wier. |
| 21:15.56 | alfonso_ | need help with my new TDM400P w/4 FXO mods. |
| 21:15.57 | Essobi | Wierd. |
| 21:16.01 | terrapen | and now they leave my business cold |
| 21:16.03 | CosmicRay | terrapen: there is a phone number on their page. |
| 21:16.14 | terrapen | cosmic: good luck on getting someone to answer it |
| 21:16.27 | Essobi | terrapen JerJer owns NuPhone. ;) |
| 21:16.34 | CosmicRay | terrapen: are you using their DID service? |
| 21:16.36 | Essobi | nufone even |
| 21:16.41 | heison | alfonso_: changing to qualify=2000 made no difference, i'm still getting app_dial.c:936 dial_exec_full: unable to create channel error |
| 21:16.41 | Essobi | I can't fricking type today. |
| 21:16.42 | terrapen | esso: i know |
| 21:16.45 | terrapen | cosmic: yup |
| 21:16.47 | bacondoublechz | anyone know how to get the callerid blocking option in CISCO phones to work with * |
| 21:16.48 | xantus | and he has a bad price to chile mobile phones |
| 21:16.54 | CosmicRay | terrapen: ahh. you are screwed then |
| 21:16.57 | terrapen | heh |
| 21:17.07 | terrapen | i paypaled them funds yesterday |
| 21:17.08 | heison | Essobi: can you show me your SIPDefault.cnf? |
| 21:17.15 | terrapen | my account balance is running dry |
| 21:17.19 | Essobi | I don't have anything special in mine bub. |
| 21:17.28 | Essobi | I only run two 60's on sip. |
| 21:17.32 | Sedorox | CosmicRay: very interesting background on the page.. |
| 21:18.12 | xantus | seen jerjer[mobile] |
| 21:18.13 | heison | Essobi: while i was upgrading firmware i lost my SIPDefault.cnf, and i suspect it has something to do with that... coz the working phone has not been rebooted |
| 21:18.19 | xantus | wtf, no bot/ |
| 21:18.27 | heison | ~seen JerJer[mobile] |
| 21:18.29 | jbot | jerjer[mobile] <~jj@feth100-fw.fament.net> was last seen on IRC in channel #asterisk, 22h 50m 57s ago, saying: 'and we absolutely do not use the word unlimited'. |
| 21:18.45 | xantus | ~seen jerjer |
| 21:18.46 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 12d 4h 59m 51s ago, saying: 'mrgoby: sure'. |
| 21:18.49 | terrapen | i should look at TxLink |
| 21:18.54 | terrapen | we use Texlink for our T1s |
| 21:19.01 | terrapen | should give em a try for VoIP |
| 21:19.06 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 21:19.18 | heison | is anyone here kind enough to show me his SIPDefault.cnf? |
| 21:19.29 | Essobi | lemme look |
| 21:19.31 | bacondoublechz | anyone? callerid blocking +cisco +sip? |
| 21:19.41 | *** join/#asterisk tessier (~treed@210.245.99.140) |
| 21:19.45 | heison | i'm missing all the new stuff after Release 4.0 |
| 21:20.10 | Essobi | Here |
| 21:20.17 | xantus | i dial *67 before dialing a number to block cid |
| 21:20.26 | terrapen | i love this wiki news item: |
| 21:20.27 | terrapen | <PROTECTED> |
| 21:20.46 | xantus | haha |
| 21:20.47 | terrapen | i, for one, welcome our non-vonage news item overlords. |
| 21:20.53 | bacondoublechz | xantus, I was taking about the option in the call preferences menu |
| 21:21.30 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 21:21.31 | lesouvage | I read some posts and info about the sipura 3000. It doesn't seem to be a plug and play device. Is it really that hard to get it working with one phone line, sipura 3000 and the asterisk box with all the features. |
| 21:23.01 | *** part/#asterisk roamer323 (~sing@70.50.113.2) |
| 21:23.09 | Ash | lesouvage: a friend who just got a sipura 3000 had a heck of a time getting it all working |
| 21:23.15 | Ash | but it's apparently working fine now |
| 21:23.19 | Ash | just took him a day or two |
| 21:23.22 | heison | the working 7960 has date and time at the top, the non-working one doesn't have date/time |
| 21:23.41 | xstaci | anyone have any ideas why channels remain open after i've hung up? i have like 10 channels open that were supposedly 'hung up' |
| 21:23.48 | heison | when i say working and non-working... it actually refers outgoing calls only on the non-working phone |
| 21:24.12 | Ash | heison: just snag a new SIPDefault.cnf from cisco |
| 21:24.24 | heison | ash: i just did |
| 21:24.30 | Ash | heison: great! |
| 21:24.32 | Ash | hehe |
| 21:24.37 | __cmc__ | do most folks install from scratch or use something like *@home ? |
| 21:24.47 | Ash | I installed from scratch |
| 21:24.51 | Ash | but I can't speak for anyone else |
| 21:24.58 | heison | ash: but it still doesn't seem to work, that why i asked essobi for his |
| 21:24.59 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 21:25.03 | SexyKen | Why would people use Asterisek at home? |
| 21:25.08 | Ash | heison: do you have a SIPmacaddr.cnf ? |
| 21:25.13 | heison | yes |
| 21:25.15 | MattH | Hi... couple of questions about askerisk..... #1 Can I setup more than 1 queue? |
| 21:25.25 | Ash | heison: does it have the right settings in it? |
| 21:25.32 | Ash | MattH: what do you mean by 'queue' ? |
| 21:25.46 | heison | ash: yes, i did a diff before and after the hardware swap |
| 21:25.53 | heison | they are identical |
| 21:25.58 | __cmc__ | one possibly nice thing I like about *@home is that is comes with AMP and Flash Operator Panel pre-installed and configured |
| 21:26.04 | Ash | wait, you swapped hardware? |
| 21:26.05 | MattH | sorry... rather I guess I should ask.. CAN asterisk do queuing? |
| 21:26.10 | MattH | like all agents are busy.. please hold? |
| 21:26.13 | Ash | the new hardware should have a different mac address |
| 21:26.18 | Ash | MattH: sure |
| 21:26.20 | sambal | MattH: yes |
| 21:26.41 | MattH | what is that under? |
| 21:27.05 | heison | ash: yes the new hardware has different mac address, and i updated dhcpd.conf to reflect that, and i also created a new SIPmac.cnf for the new hardware and compare between SIPoldmac.cnf and SIPmac.cnf - they are identical |
| 21:27.10 | Ash | MattH: you'd have to set it up. |
| 21:27.20 | Ash | heison: and the new one is running the same version of firmware? |
| 21:27.38 | MattH | ash: understood.. but what would it be setup under?? |
| 21:27.55 | Ash | MattH: You'd have to do that in extensions.conf |
| 21:27.55 | heison | it came with SCCP, and I went thru the pain of upgrading to SIP 7.3 |
| 21:28.01 | Ash | ah |
| 21:28.02 | heison | same firmware as before |
| 21:28.08 | Ash | weird |
| 21:28.14 | Ash | what's the phone doing? nothing? |
| 21:28.29 | heison | ash: i can dial out, just not accepting calls... |
| 21:28.47 | Ash | heison: at the asterisk console is it showing up as registered? |
| 21:28.49 | Ash | sip show peers |
| 21:28.50 | heison | i thought i might be CODEC related and so i went and change sip.conf and do allow=all |
| 21:29.17 | Ash | turn on sip debugging at the cli and try calling your phone, heison |
| 21:29.19 | heison | ash: ah, cisco1/cisco1 (Unspecified) D 255.255.255.255 0 UNKNOWN |
| 21:29.26 | Ash | yeah |
| 21:29.28 | Ash | :) |
| 21:29.38 | heison | what might be the problem? |
| 21:29.44 | *** part/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 21:31.18 | Ash | heison: is your phone set up to register in SIPmacaddr.cnf? |
| 21:31.25 | SPoon_TSX | Hi there, just wondering where did you guy buy your VoIP equitment off? |
| 21:31.31 | Ash | also check to see if you have the proper definitions in sip.conf on the asterisk side |
| 21:31.48 | Ash | gotta run to lunch, bbiaf |
| 21:32.05 | Ash | SPoon_TSX: voipsupply.com is where we bought all our stuff. |
| 21:32.32 | PTG123 | hey would i be able to find an adapter for a cisco 7960 locally? anyone know? :) |
| 21:32.49 | SPoon_TSX | How is their service? Are they delievery their package on time? |
| 21:32.58 | heison | ash: in SIPmacaddr.cnf i have the name, shortname, displayname, authname & password define which match sip.conf |
| 21:33.03 | lesouvage | __cmc__: I installed xorcom. Its easy to install (just put the cd in and have a nice cup of coffee) and it's ready to use. You can use it as a start. I run it on vmware for learning and testing. |
| 21:33.04 | Ash | service was quite fine |
| 21:33.06 | spackle | SPoon_TSX: we bought stome stuff from VOIPsupply, Asteriskmall.com and PCConnection. |
| 21:33.08 | Ash | sorry, i'm outta here |
| 21:33.09 | Ash | bbiaf |
| 21:33.30 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 21:33.37 | heison | i'm rebooting the working phone to see if it is related to tftp server |
| 21:33.48 | xstaci | im having problems with calls not being 'hung up' and leaving channels open. can anyone help? |
| 21:33.49 | spackle | Spoon_TSX: voipsupply has excellent service, they are really set up for the long haul. |
| 21:34.42 | `Sauron | CosmicRay: changelog is yours? |
| 21:36.05 | Pinhole | I want to play games on my sip phone. Anybody have anything cool (or cool ideas) ? |
| 21:36.47 | Pinhole | I have "I'm thinking of a number" and "rock, paper, scissors" |
| 21:37.04 | heison | looks like something to do with tftp server, coz after rebooting the working phone, i lost the date/time stamp |
| 21:37.37 | heison | but i can still call from/to the phone that just rebooted... |
| 21:38.25 | Essobi | Pinhole The umm.. "Math game" |
| 21:38.27 | Essobi | Heh. |
| 21:39.19 | Pinhole | math game? |
| 21:39.27 | *** join/#asterisk _570RM_ (~storm@80.172.149.171) |
| 21:39.29 | Essobi | 1 + 1 is.... |
| 21:39.32 | *** join/#asterisk tuxinator_linuxM (~tuxinator@m090e36d0.tmodns.net) |
| 21:39.37 | Essobi | 2! WEEEEEE you're right. |
| 21:39.44 | Essobi | someone wrote an AGI for it. |
| 21:39.46 | Essobi | Heh. |
| 21:39.53 | Essobi | in perl I think. |
| 21:39.56 | puppet | Anyone more then me that got probs with Moh in 1.0.6? |
| 21:40.17 | Essobi | Umm. |
| 21:40.18 | Essobi | Yea |
| 21:40.23 | Essobi | there is an MOH patch for it |
| 21:40.31 | puppet | oh |
| 21:40.32 | Essobi | 1.0.6 MOH is BUSTED. |
| 21:40.34 | Essobi | Heh. |
| 21:40.45 | puppet | i thought it was my computer all the time |
| 21:40.46 | puppet | lol |
| 21:40.47 | Pinhole | ~MOH |
| 21:40.50 | tuxinator_linuxM | have you checked Mantis? |
| 21:41.01 | tuxinator_linuxM | ~mantis |
| 21:41.02 | jbot | hmm... mantis is the opie bugtracker located at http://opie.info or something that tille says has a life of its own. at http://www.opie.info/mantis/ |
| 21:41.07 | tuxinator_linuxM | hmm |
| 21:41.55 | tuxinator_linuxM | jobi: Matis is at http://bugs.digium.com |
| 21:42.56 | tuxinator_linuxM | jbot: Mantis is at http://bugs.digium.com |
| 21:42.57 | jbot | ...but mantis is already something else... |
| 21:43.10 | tuxinator_linuxM | jbot: Mantis is at http://bugs.digium.com |
| 21:43.11 | jbot | ...but mantis is already something else... |
| 21:43.17 | Sedorox | jbot: forget mantis |
| 21:43.17 | jbot | Sedorox: i forgot mantis |
| 21:43.28 | Sedorox | jbot: Mantis is at http://bugs.digium.com |
| 21:43.30 | jbot | Sedorox: okay |
| 21:43.35 | tuxinator_linuxM | thanks, Sedorox |
| 21:43.38 | Sedorox | yup |
| 21:43.39 | tuxinator_linuxM | I forgot how |
| 21:43.45 | puppet | hmmm |
| 21:43.50 | tuxinator_linuxM | I am so tired |
| 21:43.50 | Sedorox | we have a bot similiar in the computer club's channel |
| 21:43.51 | Sedorox | so.. |
| 21:43.57 | puppet | *trying to find moh patch* |
| 21:43.58 | sudhir492 | I cannot seem to use both ports of PAP2-NA at the same time. When I call both at the same time, I see on the second line: Got SIP response 488 "Not Acceptable Here" back from 203.197.53.81 |
| 21:45.23 | tuxinator_linuxM | puppet: maybe http://bugs.digium.com/bug_view_page.php?bug_id=0003565 |
| 21:45.44 | puppet | searched for music didnt find ;) |
| 21:45.50 | opus___ | jbot: hello |
| 21:45.51 | jbot | Howdy Bub |
| 21:45.52 | puppet | thanks |
| 21:45.58 | opus___ | jbot: what time is it |
| 21:46.00 | jbot | opus___: I think you lost me on that one |
| 21:46.06 | opus___ | jbot: say hello |
| 21:46.08 | jbot | hello |
| 21:46.12 | opus___ | jbot: say `date` |
| 21:46.14 | jbot | `date` |
| 21:46.34 | opus___ | jbot: say jbot: say jbot: hello |
| 21:46.36 | jbot | jbot: say jbot: hello |
| 21:46.37 | tuxinator_linuxM | puppet: It is fixed in 1.0.7 |
| 21:46.54 | puppet | hmm i can just wait for .7 then |
| 21:47.04 | tuxinator_linuxM | It is relased |
| 21:47.09 | tuxinator_linuxM | http://bugs.digium.com/bug_view_page.php?bug_id=0003746 |
| 21:47.14 | puppet | as cvs |
| 21:47.37 | puppet | or did i miss a link |
| 21:47.59 | tuxinator_linuxM | in is in cvs |
| 21:49.50 | puppet | i hate cvs ;D ill wait ;D |
| 21:53.10 | coldfeet | numerical values for extensions, can this be altered to handle names eg. exten =>jack,1,hangup etc, I am trying to route to voicemail from SER, and the sip address usually has a name on it, so rather than calling another translation somewhere how hv other implemented this |
| 21:53.42 | john867530 | Anyone know of any frontends for meetme that aren't out of date? |
| 21:53.42 | heison | ash: i have put back the old phone, and the problem persist |
| 21:53.45 | bannerman | I was wondering that too, coldfeet |
| 21:53.51 | *** join/#asterisk _tommyg_ (~tom@vsat-148-64-73-166.c119.t7.mrt.starband.net) |
| 21:54.03 | Essobi | john I hacked up that one that uses mysql |
| 21:54.46 | john867530 | Essobi, That is exactly what I am looking for |
| 21:54.53 | john867530 | do you have it avalible on the net somewhere |
| 21:54.58 | Essobi | Nope. |
| 21:55.15 | Essobi | It's not as full-fledged as the current * one |
| 21:55.34 | Essobi | but it works.. and I hacked up some of the MySQL code to suite my needs.. |
| 21:55.39 | coldfeet | banner the only way i can think is b4 its passed from ser to asterisk that the numerical aliases are called , but thats a pain |
| 21:55.42 | Essobi | ANI and DNIS display.. etc. |
| 21:55.45 | john867530 | well I know that meeetme2 uses mysql but the code is out of date. |
| 21:56.00 | Essobi | That's what I hacked up into -head. |
| 21:56.19 | Essobi | took about 30 minutes to fix it and to get the code I wanted into it too |
| 21:56.47 | Nugget | meetme2 uses mysql? oh jesus, whatever for? |
| 21:57.02 | Essobi | PHP conference management. |
| 21:57.09 | john867530 | is there anyway I could get the patch ?!?! I am not a C programmer I have spent 2 nights trying to hack the -head |
| 21:57.10 | Nugget | that sucks |
| 21:57.37 | coldfeet | in fact I think its possible using goto |
| 21:58.16 | coldfeet | let me try it |
| 21:58.31 | john867530 | oh ok I understand that thanks anyway |
| 21:58.58 | rvhi | try this, in ser add an prefix to the ruri, e.g. 234@test.com would become vm+234@test.com |
| 21:58.58 | sudhir492 | Anyone using SPA-2000 or PAP2-NA here? |
| 21:59.06 | rvhi | then route it to voicemail |
| 22:00.01 | *** join/#asterisk footnote (~jhicks@67.141.135.121) |
| 22:00.23 | john867530 | the code for meetme2 is close to meetme.c but I think the variable names are different |
| 22:02.18 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 22:02.52 | *** join/#asterisk bah (048830696@AC8F8B8D.ipt.aol.com) |
| 22:02.55 | heison | does anyone know what this means: Mar 11 17:02:36 WARNING[1381]: chan_sip.c:650 __sip_xmit: sip_xmit of 0x815706c (len 411) to 10.155.10.5 returned -1: Invalid argument |
| 22:03.26 | opus___ | hmmm. i have no idea but i want to guess |
| 22:03.28 | JerJer[mobile] | heison: is that the proper IP addy for that machine? |
| 22:03.36 | heison | 10.155.10.6 |
| 22:03.49 | JerJer[mobile] | hmm |
| 22:03.52 | heison | sorry, i meant 10.155.10.6 |
| 22:03.55 | *** join/#asterisk eKo1 (~bernd@207.42.191.66) |
| 22:04.06 | opus___ | my guess is that a udp packet came in corrupted that caused chan_sip/__sip_xmit() to throw an exception. |
| 22:04.07 | JunK-Y | whats the difference now? |
| 22:04.49 | heison | JerJer: i started getting this weird problem after I tried to swap hardware (RMA returned 7960)... I can make outgoing calls, but cannot receive calls |
| 22:04.54 | Moc____ | damn busy day |
| 22:05.05 | heison | sip show peers returns |
| 22:05.10 | *** join/#asterisk DannyF (~wizardone@h133n4c2o848.bredband.skanova.com) |
| 22:05.12 | opus___ | heison - what does 'sip show registry' show |
| 22:05.14 | heison | cisco2/cisco2 10.155.10.9 D 255.255.255.255 5060 Unmonitored |
| 22:05.14 | heison | cisco1/cisco1 (Unspecified) D 255.255.255.255 0 UNKNOWN |
| 22:05.18 | DannyF | lo folks |
| 22:05.29 | heison | cisco1 is the one that can make calls but not receive |
| 22:05.33 | heison | cisco2 can do both |
| 22:05.58 | heison | cisco3 (currently not online) was swapped with cisco1 |
| 22:06.03 | opus___ | sip show registry ? |
| 22:06.16 | heison | sip show registry |
| 22:06.16 | heison | Host Username Refresh State |
| 22:06.16 | heison | fwd.pulver.com:5060 218889 105 Registered |
| 22:07.03 | opus___ | nevermind |
| 22:07.17 | Moc____ | JunK-y, so did it work ? |
| 22:07.21 | opus___ | what is the network setup as |
| 22:07.26 | heison | i tried rebooting *, cisco1, cisco2 & cisco3, swap cisco1 and cisco3 again... i'm still experiencing the same problem |
| 22:07.32 | heison | 10.155/16 |
| 22:07.37 | heison | flat |
| 22:07.42 | DannyF | anyone on that has played with the app_nv_faxdetect apps? works like a charm... but seams to force codec to slin... |
| 22:07.43 | heison | off 3548 |
| 22:07.54 | heison | what's interesting is... |
| 22:08.50 | heison | when i reboot cisco2 (and boot off tftp), and so does cisco1; i no longer have the date/time display on the phone |
| 22:09.05 | opus___ | have you tried rebooting the switch then each device and then asterisk, in that order? |
| 22:09.11 | opus___ | switch being the network switch |
| 22:09.19 | heison | yes |
| 22:09.43 | *** join/#asterisk Dark40rce (~vince@dsl-17-106.cofs.net) |
| 22:10.45 | heison | at first, i suspect it's problem on the new 7960, but when i swap it back the problem persist |
| 22:10.58 | heison | telling me it's either Asterisk, the network or the tftp server |
| 22:10.59 | eKo1 | What problem? |
| 22:12.51 | FuriousGeorge | i used the xorg script to configure x. my mouse wheel doenst work so i look in xorg.conf to see why. the entrie mouse section is commented out. the mouse is usb |
| 22:12.59 | PTG123 | Does anyone know how to rest admin password on the polycom phones? |
| 22:13.14 | FuriousGeorge | i selected auto when xorg.conf asked me about the mouse |
| 22:13.16 | heison | eKo1: i have 2 7960's (cisco1 & cisco2), both work perfectly fine until i swapped hardware (cisco1 with cisco3). cisco3 can only make outgoing calls but no accept incoming, asterisk complains about cause 3 (no route). So I swapped cisco3 and cisco1 again, the problem follows the extension, I cannot receive calls but i can make calls |
| 22:13.40 | FuriousGeorge | whoops, im in the wrong wroom |
| 22:13.42 | FuriousGeorge | pardon mje |
| 22:14.24 | Moc____ | PTG123: urm, Can you ? |
| 22:14.44 | opus___ | does a tcpdump of asterisk <-> not working phone show that it is sending packets to it to ring? |
| 22:15.06 | heison | when i swap hardware, i changed dhcpd.conf and created copy SIPmacaddr.cnf to SIPnewmacaddr.cnf |
| 22:15.09 | *** join/#asterisk GDRA (~1054@ool-18b9dba0.dyn.optonline.net) |
| 22:15.16 | Moc____ | opus, you might need to increate the packet size accepted thought |
| 22:15.19 | *** join/#asterisk onixx (1000@CPE0040f47145d1-CM000f9f7f2290.cpe.net.cable.rogers.com) |
| 22:16.06 | onixx | hi all, I'm currently a nufone customer, anything else interesting for outbound only ? |
| 22:16.11 | *** join/#asterisk jeffik (~jeffik@user-v7kajrp.dialup.mindspring.com) |
| 22:16.21 | heison | [root@pbx asterisk]# tcpdump -n -i eth0 host cisco1 |
| 22:16.21 | heison | tcpdump: listening on eth0 |
| 22:16.21 | heison | <PROTECTED> |
| 22:16.57 | opus___ | try rebooting the phone and verify network traffic |
| 22:18.26 | heison | i can see traffic when i hit the message button on the 7960 |
| 22:19.07 | MikeJ_ | it is not registerd to *, so you will see nothing when you try to call it.. |
| 22:19.31 | sivana | is the CVS working for anyone right now? |
| 22:19.34 | opus___ | hmmm.. i would double, triple check your sip.conf, even to the point of printing it out and retyping it in. |
| 22:20.00 | Nugget | cvs is bork. |
| 22:20.07 | Nugget | s root@suburbia:/usr/src/asterisk#make update 16:19:51 |
| 22:20.11 | Nugget | Updating from CVS... |
| 22:20.14 | Nugget | can't create temporary directory /tmp/cvs-serv23445 |
| 22:20.15 | JerJer[mobile] | define bork |
| 22:20.16 | Nugget | Disk quota exceeded |
| 22:20.23 | dolson | if I set my codec's ptime from 20ms up to 40ms, will that help with voice quality issues? |
| 22:20.28 | sivana | Nugget: is that french for broke? |
| 22:20.31 | MikeJ_ | which cvs ip is dead? |
| 22:20.51 | Nugget | dunno, it doesn't say which IP I got just there |
| 22:21.53 | onixx | anyone know if nufone allow multiple outbound @ the same time with one account |
| 22:21.59 | MikeJ_ | it'll be in your dns cache |
| 22:22.12 | JerJer[mobile] | onixx: yes |
| 22:22.15 | Nugget | no, my dns cache has all the rr ips |
| 22:22.25 | JerJer[mobile] | its not my mirror |
| 22:22.35 | JerJer[mobile] | is bkw still the other mirror? |
| 22:22.42 | onixx | JerJer[mobile]: thanks for the answer, I guess I'll do business with them then |
| 22:22.45 | sivana | JerJer[mobile]: you have a CVS mirror? |
| 22:22.53 | sivana | can I use it plz? |
| 22:23.08 | JerJer[mobile] | cvs.digium.com |
| 22:23.16 | sivana | ah.. its broke |
| 22:23.26 | JerJer[mobile] | just do it again |
| 22:23.32 | JerJer[mobile] | round robbin |
| 22:23.35 | Nugget | there are two servers in the round-robin for cvs.digium.com |
| 22:24.00 | sivana | [root@prepaid-1 src]# cvs co asterisk |
| 22:24.00 | sivana | can't create temporary directory /tmp/cvs-serv23841 |
| 22:24.00 | sivana | Disk quota exceeded |
| 22:24.07 | Nugget | just do it again. |
| 22:24.11 | JerJer[mobile] | try agan |
| 22:24.11 | Nugget | 50% of the time it will work. :) |
| 22:24.16 | sivana | oh.. just keep trying? |
| 22:24.40 | sivana | gotcha.. it hit a working one :) |
| 22:24.42 | MikeJ_ | there are multiple cvs servers... jerjer runs one.. it is not his that is dead... we need to know which one to get it fixed |
| 22:24.50 | Nugget | there's only one other one. |
| 22:24.55 | Nugget | so it should be simple to deduce |
| 22:24.56 | MikeJ_ | bkw is still in california.. |
| 22:25.06 | *** join/#asterisk DaLion (DaLion@Toronto-HSE-ppp3770824.sympatico.ca) |
| 22:25.15 | DaLion | hey VC pick the damn phone |
| 22:25.23 | sivana | the one I hit is slower than normal :) |
| 22:25.38 | Nugget | "working" is infinitely faster than "broken", though. |
| 22:25.44 | sivana | yup :) |
| 22:26.00 | `Sauron | Rawr |
| 22:26.04 | `Sauron | what up dalion? |
| 22:26.35 | *** join/#asterisk WhiteWlf (WhiteWolf@CPE-69-76-133-249.kc.rr.com) |
| 22:27.17 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
| 22:28.03 | sivana | thanks guys for the cvs tip :) |
| 22:28.28 | *** join/#asterisk pauldy (jeffy@24-155-82-32.ip.grandenetworks.net) |
| 22:28.55 | pauldy | anyone have any resources identifying current broadvoice/asterisk configurations? |
| 22:29.13 | pauldy | I have everything working but outbound sip calls have no audio |
| 22:29.30 | AgiNamu | Has anyone here used G711 with RGL compression? |
| 22:29.31 | bonez39 | pauldy: I have to ask...what is then an outbound call, sans audio? |
| 22:29.36 | bonez39 | not really a call is it? |
| 22:29.37 | MicH323 | mine BV is working |
| 22:29.37 | dolson | no one knows |
| 22:29.41 | AgiNamu | It saves on average ~10% of bandwidth |
| 22:29.46 | AgiNamu | well, frame size |
| 22:29.53 | DaLion | ahahahahaha |
| 22:29.53 | pauldy | means it rings the remote end but when you pick up you can't hear them they can't hear you |
| 22:30.00 | AgiNamu | but it's lossless. |
| 22:30.07 | DaLion | i guess VC a LVoip HAVE zero service |
| 22:30.11 | Hmmhesays | lookin' for a lova in world that's much to dark, because you don't no no, you want my satisfaction |
| 22:30.13 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 22:30.17 | AgiNamu | One way audio? sounds like SIP is sucking again! |
| 22:30.34 | MikeJ_ | that would be sip is sucking still |
| 22:30.40 | *** join/#asterisk Goldenear (~goldenear@d193.dhcp212-198-200.noos.fr) |
| 22:30.44 | pauldy | works ok if you use the call manager from broadvoice |
| 22:30.53 | pauldy | but thats hardly practical |
| 22:31.14 | pauldy | thought it was a codec issue but now I thikn it is in the registration |
| 22:31.22 | pauldy | for the outbound call |
| 22:31.48 | AgiNamu | MAybe with SGI doing asterisk |
| 22:31.48 | MikeJ_ | registration for an outbound calll? |
| 22:31.53 | AgiNamu | we'll see some more pickup on IAX? |
| 22:31.57 | MicH323 | pauldy: Is your codec synced |
| 22:32.13 | opus___ | h323 - how can you tell? |
| 22:32.18 | pauldy | I am using the same codec for outbound calls as I am using for inbound |
| 22:32.23 | pauldy | ulaw |
| 22:32.39 | pauldy | not sure if it is synched or not |
| 22:33.06 | Goldenear | Is IAX suitable for Presence and Instant Messaging ? |
| 22:33.12 | MicH323 | ulaw is a safe bet! |
| 22:33.16 | sudhir492 | JerJer[mobile]: Is it possible to use both the phone ports of SPA-2000 or PAP2-NA at the same time? |
| 22:33.22 | Nivex | Goldenear: Firefly seems to think so |
| 22:33.33 | MikeJ_ | sudhir. yes, you can use both at once |
| 22:33.38 | Nivex | Goldenear: but I think they use some proprietary extensions |
| 22:33.46 | AgiNamu | Yea |
| 22:34.18 | Goldenear | Nivex: yep I believe Firefly use their extended IAX to do IM |
| 22:34.40 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 22:34.58 | Goldenear | But the "stantard" IAX isn't ready for that, isn't it ? |
| 22:35.03 | pauldy | MicH323 do you mean I don't need to worry about synch with ulaw? |
| 22:35.12 | AgiNamu | goldenear, well, there's no software |
| 22:35.13 | AgiNamu | so |
| 22:35.48 | sudhir492 | MikeJ_: I get SIP response 488 back. Is there some magic in setting? |
| 22:35.56 | Juggie | jbot seen Cresl1n |
| 22:35.58 | jbot | cresl1n <~matt@216.207.245.23> was last seen on IRC in channel #asterisk, 7d 6h 20m 57s ago, saying: '~~'. |
| 22:36.24 | sudhir492 | My asterisk version is quite new: Asterisk CVS-HEAD-02/27 |
| 22:36.35 | Goldenear | AgiNamu: is it possible to make a such software (IM) without tweaking IAX (and using the standard IAXclient lib) ? |
| 22:37.19 | pauldy | I get this error with sip debug "481 Call/Transaction Does Not Exist" |
| 22:37.19 | DaLion | Voiceconduits NEVER ASNWER .. plus signup FUCKED... never works.. 80% of theyre site Is SHIT |
| 22:37.31 | eKo1 | Goldenear: You want to make an IM client/server using IAX? |
| 22:38.00 | DaLion | man,.. plus AIM msn etc not working ... |
| 22:38.25 | Goldenear | eKo1: yes, I'm thinking about it ... |
| 22:38.35 | eKo1 | <PROTECTED> |
| 22:38.47 | AgiNamu | Yea, i think so |
| 22:38.52 | AgiNamu | I havent studied the iax protocol much |
| 22:38.59 | AgiNamu | so i dont really know. But if FireFly did it |
| 22:39.08 | AgiNamu | It'd be cool if Virbiage actually had humans there |
| 22:39.12 | AgiNamu | that could answer |
| 22:39.16 | AgiNamu | Well, screw that |
| 22:39.19 | AgiNamu | break out Etheral |
| 22:39.25 | AgiNamu | ethereal, and figure out how they did it |
| 22:39.35 | Goldenear | but Firefly is not open source AFAIK |
| 22:39.39 | eKo1 | Somebody was bugging kram about how IAX works and writing drafts/white papers on it. |
| 22:39.55 | eKo1 | Who was that? |
| 22:39.56 | AgiNamu | that's why you use ethereal |
| 22:40.01 | AgiNamu | I WAS at VON last year |
| 22:40.14 | DaLion | lol |
| 22:40.16 | AgiNamu | I said "before you implement encryption, *write a paper* describing how you plan on doing it" |
| 22:40.19 | DaLion | livevoip just called me |
| 22:40.30 | DaLion | seems.. they dont have canada dids no more |
| 22:40.33 | DaLion | lol |
| 22:40.36 | AgiNamu | course, why the hell should anyone bother to spec out something as unimportant as security eh? |
| 22:40.47 | AgiNamu | We're all better off without threat models and all those useless docs. |
| 22:40.52 | DaLion | ok anyone from VC here ? |
| 22:40.59 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 22:41.00 | DaLion | so i can tell them theyre site sucks.. |
| 22:41.01 | ManxPower | !docs |
| 22:41.08 | ManxPower | ~docs |
| 22:41.09 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 22:41.28 | DaLion | `Sauron.. what was it u answered me ? |
| 22:41.47 | AgiNamu | VC? |
| 22:41.49 | `Sauron | nothing |
| 22:41.50 | AgiNamu | VoiceCall? |
| 22:41.50 | `Sauron | saying hi |
| 22:41.53 | AgiNamu | VoipCenter? |
| 22:42.00 | eKo1 | vancouver? |
| 22:42.05 | AgiNamu | haha |
| 22:42.10 | AgiNamu | Very Cold. |
| 22:42.13 | DaLion | ah ..hehe |
| 22:42.13 | Goldenear | eKo1: www.cornfed.com/iax.pdf <-- is it what you mean ? |
| 22:42.13 | DaLion | so anyone know where to get CANADA dids.. central or eastern.. 418 area code |
| 22:42.20 | AgiNamu | 418? lemme check |
| 22:42.24 | eKo1 | Goldenear: yeah |
| 22:42.26 | DaLion | VC= voiceconduits |
| 22:42.43 | CosmicRay | DaLion: did you check http://www.voip-info.org/wiki-VOIP+Service+Providers+Residential and related pages? |
| 22:42.47 | *** join/#asterisk brettnem (~mive29@user-0ccsr2l.cable.mindspring.com) |
| 22:42.58 | CosmicRay | DaLion: only about two dozen canadian companies listed :-) |
| 22:43.04 | puppet | this sucks |
| 22:43.05 | puppet | 64 bytes from sipgw (83.140.41.62): icmp_seq=3 ttl=57 time=1194 ms |
| 22:43.08 | CosmicRay | including several that explicitly list did |
| 22:43.08 | puppet | to my sipgw... |
| 22:43.25 | AgiNamu | Sorry. I only have afew in BC, Ontario, and Alberta |
| 22:43.47 | Goldenear | eKo1: the document only refer to a text (raw text) as a possible media ... but no details about presence ... :( |
| 22:43.51 | sivana | heh |
| 22:43.51 | *** part/#asterisk BurnedOutGeek (~BurnedOut@216.215.202.4.nw.nuvox.net) |
| 22:43.52 | pauldy | !broadvoice |
| 22:44.16 | sivana | DaLion: whatcha looking for? |
| 22:44.21 | DaLion | yes i checked |
| 22:44.36 | Delmar | anyone here using X100P cards please? |
| 22:44.41 | AgiNamu | Goldenear, well, send send raw text |
| 22:44.42 | AgiNamu | with headers |
| 22:44.49 | AgiNamu | IAX won't support all the headers you need anyways |
| 22:44.50 | tclark | AgiNamu: 250 in victoria BC ? |
| 22:44.56 | tuxinator_linuxM | puppet: Minimum = 672ms, Maximum = 1075ms, Average = 799ms |
| 22:45.14 | puppet | tuxinator_linuxm: our connection suck :( |
| 22:45.17 | tuxinator_linuxM | from me using tmobile aircard |
| 22:45.27 | AgiNamu | 250-483 (BC VICTORIA ) |
| 22:46.08 | *** join/#asterisk Markie (Mark@208.247.106.165) |
| 22:46.32 | Markie | i think form now on, we all just talk DTMF in here |
| 22:46.37 | Markie | **##<flash> |
| 22:46.46 | DaLion | basically need IAX to my * ... not some shitty vonage ATA's |
| 22:46.52 | Delmar | lol u flasher !! |
| 22:47.04 | tuxinator_linuxM | I'm at Meet * right now |
| 22:47.05 | Delmar | i mean.... |
| 22:47.15 | `Sauron | Markie? |
| 22:47.19 | lidl | a simple question: when I need a TE405P instead of a TDM400? |
| 22:47.20 | `Sauron | Is that really you? |
| 22:47.22 | Markie | sauron? |
| 22:47.24 | Markie | yea |
| 22:47.28 | Goldenear | AgiNamu: so Jabber/XMPP + IAX should be a better solution... or sip (but the NAT traversal issues) |
| 22:47.33 | Markie | me and whitewolf were just talking about you |
| 22:47.37 | `Sauron | MARKIE! |
| 22:47.43 | `Sauron | heh |
| 22:47.45 | `Sauron | funny boy |
| 22:47.46 | Delmar | 565 968 3527437 |
| 22:47.47 | Markie | lol..i was saying that i knoew someone on #php was a big asterisk guy too |
| 22:47.49 | AgiNamu | Goldenear, what's the market for it anywahs? |
| 22:47.49 | Delmar | meh. |
| 22:47.57 | Delmar | converting letters to numbers hurts my brain |
| 22:48.03 | `Sauron | Y'all need to quit stalking me |
| 22:48.10 | Markie | Whoop! |
| 22:48.12 | Goldenear | AgiNamu: MSN users ;) |
| 22:48.19 | Markie | i'm in here ervety now and then.. |
| 22:48.19 | eKo1 | lidl: When you're going to use an E1. |
| 22:48.21 | Markie | i aint no noob |
| 22:48.26 | Markie | just noob-ish |
| 22:48.27 | AgiNamu | haha |
| 22:48.35 | AgiNamu | yea, ok , well, after you write the interop bridge |
| 22:48.40 | AgiNamu | then you can start working on a separate client :P |
| 22:48.41 | *** join/#asterisk PyroSteve (~steve@wsip-70-183-114-254.no.no.cox.net) |
| 22:48.55 | AgiNamu | MS has a LOT of stuff coming out in the presence/voip area. |
| 22:49.01 | Markie | i was telling whitewolf how my asterisk config is all modded out and souped up |
| 22:49.21 | Goldenear | isn't MS in the SIP camp ? |
| 22:49.23 | PyroSteve | Which nick in here works for VoicePulse ? |
| 22:49.30 | footnote | AgiName: ActiveX wheee! |
| 22:50.01 | Delmar | ok im running * on linux debian (sid) kernel 2.6.10 with x100p. when i do an lsmod, it shows not only the wcfxo and zaptel, but also crc_ccitt. is this normal? anyone else seen this? im just trying to work out where my problems could be coming from.. and never noticed that crc_ccitt in there in 2.4.x |
| 22:50.19 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 22:50.32 | lidl | eKo1, ok. so when I need T1 or E1 instead of an FXO module? |
| 22:50.44 | AgiNamu | Footnote, nah, MANAGED CODE! yey! |
| 22:50.56 | eKo1 | yuck... |
| 22:50.59 | footnote | eek |
| 22:51.03 | AgiNamu | yea, go mock it |
| 22:51.17 | AgiNamu | meanwhile, my productivity is at least 5 times higher than C |
| 22:51.30 | eKo1 | But your app is 5 times slower too. |
| 22:51.32 | lidl | eKo1, I mean, what are two simple different contexts for the two technologies? |
| 22:51.35 | AgiNamu | haha ye right |
| 22:51.40 | footnote | just cause you code C like a leetle gurl? |
| 22:51.43 | footnote | hehe |
| 22:51.46 | `Sauron | oooh |
| 22:51.46 | eKo1 | lol |
| 22:51.48 | `Sauron | burn |
| 22:51.49 | AgiNamu | Managed Quake II, versus Unmanaged Quake II |
| 22:51.52 | `Sauron | them are fighting words |
| 22:51.52 | Juggie | hrmm..... i'm in a intreasdting situation at work |
| 22:51.58 | footnote | j/k :) |
| 22:52.02 | Juggie | my office is willing to pay for more developement in libpri |
| 22:52.06 | AgiNamu | the Managed version ran at 90-something% of the speed of the unmanaged one |
| 22:52.13 | Juggie | implement CD/ETC on Q.SIG/DMS100 |
| 22:52.22 | Juggie | but, when we pay someone we need a deliveralbe |
| 22:52.24 | AgiNamu | which is a minor price to pay to not have to worry about a whole class of problems |
| 22:52.27 | eKo1 | lidl: eh, read about it. |
| 22:52.32 | footnote | AgiNamu: beware the hype, they said all that about DDE too |
| 22:52.36 | Juggie | however, we want the code to become part of libpri and not be a custom patch for us |
| 22:52.37 | lidl | eKo1, where? |
| 22:52.42 | Juggie | how am i gonna resolve this |
| 22:52.51 | footnote | boy, DDE *hurt* |
| 22:52.52 | eKo1 | lidl: But a book on telecomm. and networking. |
| 22:53.24 | Markie | Juggie: make a vendor branch in cvs |
| 22:53.33 | AgiNamu | footnote huh? |
| 22:53.39 | eKo1 | s/But/byu |
| 22:53.41 | footnote | hehe |
| 22:53.41 | AgiNamu | I don't remember DDE being embedded in SQL Server to write spros |
| 22:53.47 | Markie | Juggie: thats what we do for all out customizations to external projects |
| 22:53.47 | eKo1 | s/byu/buy |
| 22:53.49 | AgiNamu | I dont remember DDE replacing the Win32 API :) |
| 22:54.01 | footnote | They turned it into OLE2 |
| 22:54.02 | Markie | Juggie: that way you can still keep up with changes in the core cvs just by merging. |
| 22:54.06 | footnote | sorta |
| 22:54.11 | footnote | on PCP |
| 22:54.11 | AgiNamu | Almost every new API from MS will be managed code |
| 22:54.39 | footnote | Single source solutions aren't usually optimal. |
| 22:54.43 | DaLion | anyone know if ocx for iax ? |
| 22:55.14 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 22:55.18 | Juggie | Markie, what i need is modificationsto libpri |
| 22:55.22 | Juggie | keeping it updated would be hell. |
| 22:55.25 | lidl | eKo1, I do know what's a T1 or E1 frame, I know signalling and all the theory around it. This is because I cannot figure out why someone would need a card to handle a T1 or E1 frame, and not a simple FXO card |
| 22:55.27 | Juggie | and i want it in the cvs anyways |
| 22:55.49 | Juggie | i think i will just pay for a "report" |
| 22:55.53 | eKo1 | lidl: eh, no you don't. |
| 22:55.56 | Juggie | which really means code :) |
| 22:55.59 | Markie | Juggie: well what i'm saying is as a worst case, no, keeping it updated is not hell |
| 22:56.00 | AgiNamu | DaLion, look in the recent -biz and -dev list |
| 22:56.04 | Juggie | the deliverable will be the report :) |
| 22:56.04 | Markie | it's cake. i do it all the time. |
| 22:56.07 | AgiNamu | someone made an OCX... EuropeSIP I think. |
| 22:56.29 | Markie | of course..you get it into libpri...kudos to you |
| 22:56.48 | lidl | eKo1, I see handling a T1 frame is a task for a central point exchange and not for an end user. it's something a telco should handle not a user, no? |
| 22:56.58 | Markie | Juggie: keeping it up to date with CVS is just doing a "cvs merge" |
| 22:57.00 | sudhir492 | Can anyone help me with PAP2-NA. I cannot use both the port at the same time |
| 22:58.20 | eKo1 | lidl: What do you want to do exactly? |
| 22:58.33 | Juggie | Markie, what i would would be a patch in libpri and it would be directally intertwined. |
| 22:58.48 | Juggie | furthermore, i want to contribute it to the community |
| 22:59.08 | Markie | ok..contribute away.. |
| 22:59.09 | lidl | eKo1, understand why someone would like to handle a T1 at home or office... |
| 22:59.33 | Markie | Juggie: i assume you dont know about CVS then? CVS takes care of the intertwining |
| 22:59.39 | Markie | but whatever |
| 22:59.48 | ManxPower | Markie, There is only so much CVS can do. |
| 22:59.51 | eKo1 | lidl: to truck calls or connect to the internet. |
| 23:00.02 | Juggie | Markie, i dont want to ever have to worry about the patch applying cleanly |
| 23:00.09 | Juggie | please understand that |
| 23:00.09 | Markie | true..if you're modifiyng structures..etc |
| 23:00.24 | Markie | submit away ! |
| 23:00.31 | Juggie | i also dont want to keep something i pay for for myself |
| 23:00.35 | Juggie | i want it back into the community |
| 23:00.41 | eKo1 | Hmm...looks like my core router is working again. |
| 23:00.41 | Markie | koolio |
| 23:00.54 | ManxPower | Yehaw! It's 4:20! |
| 23:01.02 | Markie | i thought your issue was more that they werent interested in adding it to libpri |
| 23:01.02 | `Sauron | 1700 here |
| 23:01.11 | BoRiS | 1700 here |
| 23:01.15 | Markie | 6:01pm here :( |
| 23:01.27 | Markie | wait..1700 |
| 23:01.32 | Markie | -12 = 5 |
| 23:01.34 | ManxPower | Time to de-borg |
| 23:01.46 | `Sauron | yer in EST/EDT, markie? |
| 23:01.48 | Markie | sauron: i thought you were in australia |
| 23:01.49 | `Sauron | should be 1800 |
| 23:01.54 | ManxPower | I wasn't referring to the time. |
| 23:01.55 | Markie | or do i always confuse that? |
| 23:02.03 | `Sauron | hum |
| 23:02.05 | `Sauron | you confuse it |
| 23:02.07 | `Sauron | I'm in texas |
| 23:02.08 | Markie | lol |
| 23:02.12 | `Sauron | yehaw |
| 23:02.16 | ManxPower | `Sauron, Well that explains a LOT! |
| 23:02.17 | Markie | i dont know why i always think that |
| 23:02.20 | Markie | lol |
| 23:02.24 | `Sauron | and now it's time to go cook up some fajitas |
| 23:02.30 | `Sauron | Manx: I love you too. ;) |
| 23:03.13 | ManxPower | `Sauron, My significant other is from TX. |
| 23:03.23 | ManxPower | Well, one of them, at least. |
| 23:03.27 | Markie | lol |
| 23:03.29 | `Sauron | That explains a lot |
| 23:03.31 | `Sauron | :) |
| 23:03.34 | ManxPower | 8-) |
| 23:04.07 | `Sauron | I'm kinda bummed. |
| 23:04.16 | `Sauron | IPv6 removes the abillity to broadcast. |
| 23:04.17 | `Sauron | :( |
| 23:04.22 | *** join/#asterisk xyharley (~daecon@xyharley.dsl.xmission.com) |
| 23:04.40 | mikegrb | ... |
| 23:06.03 | lidl | why a tdm400 card with 4 fxo costs about 320$ and a single port t100p costs about 500$? |
| 23:06.23 | *** join/#asterisk agave-txlink (phanop@216.81.43.75) |
| 23:06.34 | agave-txlink | howdy howdy |
| 23:07.10 | `Sauron | time for fajitas |
| 23:07.11 | `Sauron | Mmmm. |
| 23:08.16 | ManxPower | lidl, The TDM400P is actually much more complex because it has to handle analog. |
| 23:08.47 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
| 23:08.59 | MicH323 | Is it worth getting a PCI DSP board to codecs in HW for high volume Asterisk PBX? Anyone reccomend one? |
| 23:09.31 | MicH323 | Does the TDM400P do codecs in HW? |
| 23:09.50 | lidl | ok thx :) |
| 23:09.54 | lidl | nite |
| 23:10.36 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 23:11.55 | MikeJ[Jayden] | ~seen xkev |
| 23:11.56 | jbot | xkev <kevin@orbit.xmission.com> was last seen on IRC in channel #asterisk, 2h 7m 38s ago, saying: 'two sets then :)'. |
| 23:12.38 | *** join/#asterisk search_learn2005 (~Miranda@209.68.139.150) |
| 23:13.02 | footnote | ~seen nugget |
| 23:13.03 | jbot | nugget is currently on #asterisk (8h 48m 50s). Has said a total of 33 messages. Is idling for 47m 25s |
| 23:13.16 | *** join/#asterisk eye69 (magnus@195.84.97.50) |
| 23:13.17 | Delmar | ok im running * on linux debian (sid) kernel 2.6.10 with x100p. when i do an lsmod, it shows not only the wcfxo and zaptel, but also crc_ccitt. is this normal? anyone else seen this? im just trying to work out where my problems could be coming from.. and never noticed that crc_ccitt in there in 2.4.x |
| 23:13.30 | search_learn2005 | Can I use Asterisk with Vonage with VOIP phones? For its PBX purposes? |
| 23:15.10 | Delmar | nvm. it looks like zaptel wants it. |
| 23:16.41 | *** join/#asterisk |Vulture| (Vulture@dhcp64-134-129-79.wbt.phx.wayport.net) |
| 23:18.17 | agave-txlink | delmar: it's normal. |
| 23:18.25 | Delmar | yep. |
| 23:18.44 | AgiNamu | MicH323, no, Zapata is based on the idea that CPUs are more powerful and should handle codecs themsleves instead of DSPs. |
| 23:18.54 | AgiNamu | And I think with an Altix system, that might be true. |
| 23:19.43 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 23:20.13 | MicH323 | Can Asterisk do call routing to multiple carriers... ie First try number with BroadVoice, If it fails try Vonage? |
| 23:20.34 | AgiNamu | Yes |
| 23:20.41 | AgiNamu | you can do pretty much anything iwth a call |
| 23:20.45 | AgiNamu | some things are easier than others :P |
| 23:20.45 | ManxPower | MicH323, Yes, but it's more complicated that it seems at first. |
| 23:21.03 | JerJer[mobile] | there isn't much asterisk cannot do or be developed to do |
| 23:21.16 | AgiNamu | the question is all about feasibility :P |
| 23:21.27 | MicH323 | Is itsimply a matter of 1XXX., 1, DIAL(SIP/Vonage) 1XXX., 2, DIAL(SIP/BV) etc |
| 23:21.41 | ManxPower | nomad_, it's not. |
| 23:21.46 | ManxPower | ..er.. |
| 23:21.49 | AgiNamu | Well, that'll redial on busy |
| 23:21.50 | ManxPower | MicH323, no, it's not. |
| 23:22.00 | ManxPower | It won't redial on busy either. |
| 23:22.12 | ManxPower | It will call the same number a second time when the far end hangs up. |
| 23:22.20 | MicH323 | ouch! |
| 23:22.32 | ManxPower | Didn't I just say it's harder to do than you might think? |
| 23:22.39 | AgiNamu | you need to check the dialstatus variable i guess |
| 23:22.46 | AgiNamu | see what the result was |
| 23:22.51 | ManxPower | Dial sets the variable $(DIALSTATUS) when the call ends. |
| 23:22.52 | AgiNamu | and GotoIf off of that |
| 23:23.02 | ManxPower | Then you have to do what AgiNamu is saying. |
| 23:23.15 | AgiNamu | And then you'll wish there was Python embedded into the dialplan :) |
| 23:23.41 | MicH323 | I do I do... |
| 23:23.47 | ManxPower | Here's a treat for anyone that wants to wade thru my entire old Asterisk web site for interesting things like exactly what MicH323 is looking for: http://www.fnords.org/~eric/asterisk/wffs.tar.gz |
| 23:23.57 | MicH323 | exten => _011N., 3, congestion() ; No answer, nothing |
| 23:23.57 | MicH323 | exten => _011N., 102, busy() ; Busy |
| 23:24.21 | AgiNamu | mich, no, you gotta gotoif off of the DIALSTATUS |
| 23:24.46 | ManxPower | You CANNOT do provider failover without using ${DIALSTATUS} |
| 23:24.48 | *** join/#asterisk ws9455 (~ws9455@adsl-68-94-10-246.dsl.rcsntx.swbell.net) |
| 23:25.11 | Juggie | if you dont use it you'll end up placing the call twice :) |
| 23:25.11 | footnote | JerJer[mobile]: asterisk can be programmed to attract beautiful wimmen?!? |
| 23:25.15 | MicH323 | At the moment you can do Voicemail if call fails, so why not 2nd provider? |
| 23:25.40 | MicH323 | exten => s,1,Dial(${BVRINGS},60,r) |
| 23:25.40 | MicH323 | exten => s,2,Voicemail,u${BVVMBOX} |
| 23:25.41 | MicH323 | exten => s,102,Voicemail,b${BVVMBOX} |
| 23:26.26 | ManxPower | MicH323, because a you don't really care of the call is busy or answered (which is what voicemail is doing), you care about things like congestion and invalid numebrs. |
| 23:26.47 | doughecka | apear == the process of a-pearing something |
| 23:26.51 | AgiNamu | Footnote, sure. Make money, then register on www.golddigger.com |
| 23:27.13 | footnote | That's all there is to it? |
| 23:27.20 | doughecka | is JerJer[mobile] currently active? |
| 23:27.20 | doughecka | :) |
| 23:27.41 | Qwell | ~seen JerJer[mobile] |
| 23:27.43 | jbot | jerjer[mobile] is currently on #asterisk (1h 25m 25s). Has said a total of 11 messages. Is idling for 6m 40s |
| 23:27.46 | MicH323 | I tae it that 102 is a special code!!! |
| 23:28.08 | rvhi | can you have two instances of * running on the same server? |
| 23:28.58 | agave-txlink | will thos do the trick for mich ? |
| 23:29.00 | agave-txlink | exten => s,1,EnumLookup(${ARG1}) |
| 23:29.00 | agave-txlink | exten => s,2,Dial(${ENUM}) |
| 23:29.00 | agave-txlink | exten => s,3,Dial(SIP/1${ARG1}@192.168.10.1) |
| 23:29.00 | agave-txlink | exten => s,4,GotoIf($[${DIALSTATUS} = "BUSY"]?9 |
| 23:29.00 | agave-txlink | exten => s,5,Dial(SIP/1${ARG1}@192.168.10.1) |
| 23:29.01 | agave-txlink | exten => s,6,GotoIf($[${DIALSTATUS} = "BUSY"]?9 |
| 23:29.03 | agave-txlink | exten => s,7,Dial(SIP/20378011${ARG1}@172.5.5.1) |
| 23:29.05 | agave-txlink | exten => s,8,Hangup |
| 23:29.06 | agave-txlink | exten => s,9,Busy |
| 23:29.08 | pauldy | wow |
| 23:29.09 | agave-txlink | exten => s,102,Dial(SIP/1${ARG1}@192.168.10.1) |
| 23:29.11 | agave-txlink | exten => s,103,GotoIf($[${DIALSTATUS} = "BUSY"]?8 |
| 23:29.12 | agave-txlink | exten => s,104,Dial(SIP/1${ARG1}@192.168.10.1) |
| 23:29.13 | doughecka | ups |
| 23:29.14 | agave-txlink | exten => s,105,GotoIf($[${DIALSTATUS} = "BUSY"]?8 |
| 23:29.15 | Juggie | NO PASTEING |
| 23:29.17 | agave-txlink | exten => s,106,Dial(SIP/20378011${ARG1}@172.5.5.1) |
| 23:29.18 | agave-txlink | exten => s,107,Hangup |
| 23:29.21 | Juggie | ARGFGH |
| 23:29.31 | agave-txlink | just trying to help someone |
| 23:29.36 | pauldy | pm |
| 23:29.38 | Juggie | www.pastebin.ca |
| 23:29.42 | Juggie | or private message |
| 23:29.50 | pauldy | don't be slow |
| 23:29.57 | agave-txlink | not sure if that code is accurate |
| 23:29.59 | Markie | ouch |
| 23:30.01 | Markie | eyes |
| 23:30.05 | Markie | hurt |
| 23:30.24 | agave-txlink | 103 and 105 should be ?9 I guess |
| 23:30.52 | MicH323 | Thanks!!! |
| 23:30.59 | *** join/#asterisk _6Flamez_ (lklk@00045a809589.click-network.com) |
| 23:31.05 | agave-txlink | mich, replace s with something like _1XXXXXXXXXX |
| 23:31.09 | agave-txlink | I have s because that's in a macro |
| 23:31.24 | MicH323 | cool! |
| 23:31.36 | Markie | Manx: why isnt the site up anymore? |
| 23:31.39 | agave-txlink | and if you don't want to use Enum, then take out the 10X stuff and the enum call |
| 23:31.42 | *** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 23:32.08 | agave-txlink | but I guess you get the idear |
| 23:32.09 | agave-txlink | :) |
| 23:32.41 | _6Flamez_ | anyone w/ QoS supported hardware router recomendations? wired or wireless isn't an issue.. |
| 23:33.06 | MicH323 | Cisco! |
| 23:33.09 | firestrm | im trying to grock the essence of IAX connections between two * boxes, i understand about 90% of it but im missing some knowlege, can someone help me fill in the blanks? |
| 23:33.19 | MicH323 | Cant get a better QoS Router than Cisco! |
| 23:33.58 | _6Flamez_ | hmmm...for SOHO users? |
| 23:34.18 | _6Flamez_ | never really think Cisco for soho *shrug* i'll look into it.. |
| 23:34.48 | _6Flamez_ | quess i should of stated that.. |
| 23:35.12 | MicH323 | I have a Cisco 827 with 4 ports FXS.... Cant get a cheaper SoHo router to connect to Asterisk |
| 23:35.24 | MicH323 | But then I also have a 1760 :) |
| 23:35.51 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
| 23:36.09 | ManxPower | Either go with Cisco or use a PC and run Linux or BSD on it to do QoS |
| 23:36.13 | firestrm | how do i set up IAX accounts on the box im calling? anyone have any good example config? |
| 23:36.46 | firestrm | ManxPower, smoothwall works good |
| 23:36.48 | *** join/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
| 23:37.20 | *** join/#asterisk yashax (~yasha_x@65.23.36.66.nw.nuvox.net) |
| 23:37.26 | doughecka | m0n0wall is t3h b0mb |
| 23:37.35 | doughecka | ooh, xchat autofixes teh... |
| 23:37.52 | firestrm | doughecka, hmm never tried it.. |
| 23:38.03 | firestrm | sounds interisting though.. |
| 23:38.05 | doughecka | its got trafficshaping |
| 23:38.07 | MicH323 | Cisco also do 1800 routers (replaceemnt for 1760) |
| 23:38.10 | doughecka | 'qos' |
| 23:38.12 | puppet | woot ;p |
| 23:38.17 | _6Flamez_ | ya i use a linux based firewall/router for my home usage, but this is for a friend.. |
| 23:38.22 | puppet | my very first webswitchboard works ;D |
| 23:38.27 | doughecka | I can be downloading bittorrent, and play games without lag... |
| 23:38.28 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 23:38.34 | puppet | think its time to protect it since it can be sploited right now ;D |
| 23:38.35 | doughecka | over crappydsl |
| 23:38.40 | doughecka | (tm) |
| 23:39.14 | ManxPower | Cisco's QoS is a maze of conflicting options and poorly documented options. |
| 23:39.21 | ManxPower | But it's incredibly flexable. |
| 23:39.38 | MicH323 | Cisco has Trafficshaping, QoS, and RTP que (to ensure those Vocie calls arnt broken), Firewall and even does SIP aware |
| 23:40.23 | _6Flamez_ | ya that cisco does look nice.. |
| 23:40.33 | firestrm | ManxPower, if i want to set up an * box to accept IAX connection, what do i have to do on the box being called? i need to set up some sort of user account right? |
| 23:40.39 | opus___ | whats rtp que? |
| 23:41.00 | MicH323 | It pritorises RTP traffic |
| 23:41.17 | ManxPower | firestrm, What you are wanting to do would take more hand holding than I want to do right now. |
| 23:42.00 | firestrm | ManxPower, hmm ok.. I allmost grasp it, but not quite.. i will keep hacking away at it maybe i will stumble upon the answer |
| 23:42.24 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
| 23:43.11 | firestrm | i take it that is not as easy as just setting up some sort of user in iax.conf |
| 23:44.32 | AgiNamu | firestrm, it is |
| 23:44.35 | AgiNamu | you create the user, set password |
| 23:44.41 | AgiNamu | set context |
| 23:44.48 | AgiNamu | and then create a context of the same name in your extensions.conf |
| 23:45.02 | AgiNamu | go on the wiki... should tkae about 10-20 minutes the first time if you dont rush |
| 23:45.33 | firestrm | AgiNamu, im at the wiki now.. allmost understand it.. but the light in my brain is still flickering.. |
| 23:46.03 | AgiNamu | you can do it! |
| 23:46.07 | AgiNamu | We believe in you! |
| 23:46.08 | firestrm | :) |
| 23:46.13 | AgiNamu | pm me |
| 23:46.16 | AgiNamu | i'll show you my config |
| 23:47.21 | pauldy | ok I think I found the rror for my outbound sip problem |
| 23:47.33 | pauldy | Unable to create channel of type 'SIP' |
| 23:48.13 | Sedorox | when I get that.. I restart asterisk and all works... dunno who |
| 23:48.14 | Sedorox | tho* |
| 23:48.52 | pauldy | hrm restarted several times and have tried cvs until yesterday when something broke the compile |
| 23:48.57 | pauldy | right now I'm using 1.06 |
| 23:49.24 | Sedorox | haven't gone up from .5 yet |
| 23:49.52 | *** join/#asterisk YoYo (YoYo@dilbert.psknet.com) |
| 23:52.15 | *** join/#asterisk outtolunc (~chatzilla@adsl-69-110-26-49.dsl.pltn13.pacbell.net) |
