| 00:01.18 | ccfiel | jas_williams: http://pastebin.ca/3079 my sip.conf |
| 00:02.07 | ccfiel | jas_williams: http://pastebin.ca/3080 my extensions.conf |
| 00:02.31 | *** join/#asterisk IQ (~iq@65-103-164-82.omah.qwest.net) |
| 00:03.29 | jas_williams | ccfiel: What number are you calling and what phone are you using to initiate the call ? |
| 00:03.51 | jero | AgiNamu, for now it's unix-only. we(or someone) have to write the win32 audio stuff for this |
| 00:04.20 | AgiNamu | oh.... too ba. |
| 00:04.23 | ccfiel | jas_williams: i tried my firefly account to dial 1001 extension |
| 00:04.26 | AgiNamu | im definately interested whenever it runs on windows |
| 00:04.41 | AgiNamu | i dont have audio on my linux machines (unless X can forward audio and i dont know about it) |
| 00:04.44 | *** join/#asterisk Umaro (~umaro@c-24-22-76-14.client.comcast.net) |
| 00:05.00 | Umaro | Hey guys, can someone point me towards documentation on what the jumpers are for on the TE410P? |
| 00:05.13 | ccfiel | jas_williams: after that error occured :( |
| 00:06.06 | *** join/#asterisk Bentley (~bentley@S01060080c8135e6a.cg.shawcable.net) |
| 00:06.30 | AgiNamu | 400 line methods... joy |
| 00:06.31 | jas_williams | ccfiel: Is phone1 registered try a sip show peers in the cli |
| 00:08.08 | ccfiel | jas_williams : http://pastebin.ca/3081 here the output.. |
| 00:09.04 | jas_williams | ccfiel: You have no sip phones successfully registered with asterisk |
| 00:09.43 | ccfiel | jas_williams : ah..ic..sorry..im very newbie in asterisk..ill try to..register one sip phone....it..:) |
| 00:10.03 | ccfiel | jas_williams : thanks you very much... |
| 00:11.37 | jas_williams | ccfiel: This is the type of output you wold expect http://pastebin.ca/3082 |
| 00:12.04 | jas_williams | ccfiel: This shows IP addresses for each registered Phone |
| 00:12.11 | Moc | hi all |
| 00:12.43 | D1ng0 | yo, there any services that offer Vancouver BC numbers for VOIP ? |
| 00:13.14 | Moc | yes |
| 00:13.20 | ccfiel | jas_williams : ok ill try to register my soft phones..thanks a lot.. |
| 00:14.44 | Moc | the polycom ip 600 display look VERY NICE |
| 00:16.07 | _chad | damn tracking down firmware is a pain |
| 00:18.54 | tzanger | D1ng0: iax.cc does |
| 00:19.16 | Moc | hi tzanger |
| 00:20.45 | jero | nite ppl |
| 00:22.16 | *** join/#asterisk wolfson (~hehe@65.174.122.198) |
| 00:22.20 | *** join/#asterisk Legend (~legend@24.244.142.134) |
| 00:22.24 | *** join/#asterisk Umaro (~umaro@c-24-22-76-14.client.comcast.net) |
| 00:23.12 | AgiNamu | i guess single character varnames are "in" eh? |
| 00:24.06 | Moc | ip 600 and ip 500 are nearly identical thought |
| 00:24.39 | Umaro | Hey guys.. having a small problem in sip.conf where if I do bindaddr = 0.0.0.0 I get bad file descriptor, but I need to bind to two ips.. how do I do that? |
| 00:24.41 | aminorex | the current fad is for double or triple of a single character, such as 'iii', 'jjj' |
| 00:24.45 | modulus_ | [nwuser@nwn ~]# uptime |
| 00:24.45 | modulus_ | <PROTECTED> |
| 00:24.47 | modulus_ | *grin* |
| 00:25.00 | Moc | Umaro i think you cant |
| 00:27.30 | Umaro | Moc: :( |
| 00:27.46 | Moc | just redirect the port to the other |
| 00:27.58 | Moc | using iptables I think |
| 00:28.07 | _chad | does anyone know if the Smartnet CDW cisco extended policy for the 7960 will allow me to get the 7940 binaries? |
| 00:28.25 | Moc | 7940 and 7960 is the same firmware |
| 00:28.58 | _chad | sweet |
| 00:28.58 | Moc | it the same device, except 7940 doesnt have the plastic buttons, but the board in there |
| 00:29.06 | Moc | in = is |
| 00:29.17 | jas_williams | Umaro: Do you have your bindaddr= in the [General] section ? |
| 00:29.32 | jas_williams | opps [general] section |
| 00:29.39 | Umaro | jas_williams : yes |
| 00:29.47 | _chad | mac, any idea how long it takes to turn around the service contract? all the posts i've read are nightmares |
| 00:29.49 | jas_williams | It should work |
| 00:29.51 | Moc | he want to bind multiple IP on 1 machine.. |
| 00:30.23 | *** join/#asterisk Cresl1n (~matt@216.207.244.186) |
| 00:30.32 | Moc | _chad, your best chance is to find it on bittorrents I guess |
| 00:30.47 | wolfson | don't buy direct from cisco, last time it took 3 months, and the police |
| 00:31.04 | *** join/#asterisk darkskiez (~mhb@host-84-9-74-147.bulldogdsl.com) |
| 00:31.14 | jas_williams | Umaro: Try it without the spaces bindaddr=0.0.0.0 |
| 00:31.18 | junk-u | lo guys |
| 00:31.18 | darkskiez | I have to have serious works with sipgate |
| 00:31.34 | darkskiez | they changed the format of their callerid |
| 00:31.50 | darkskiez | imagine my suprise when i answered the phone to realise my ex-girlfriend filter hadnt worked |
| 00:31.50 | *** join/#asterisk pbd (~pbdavidso@12.144.118.36) |
| 00:31.58 | pbd | Good evening, all. |
| 00:31.59 | Moc | hi junk-u, i received my ip 600 toda |
| 00:32.00 | Moc | y |
| 00:32.54 | pbd | Hats off to JerJer- I finally managed to make a call from Asterisk to a Cisco Callmanager v3.3 |
| 00:33.14 | Moc | yea JerJer is the man |
| 00:33.15 | Umaro | jas_williams: nope, on 1.0.3 |
| 00:33.59 | Brixius | Hello |
| 00:34.13 | junk-u | Moc: nice, how it goes ur ip600? |
| 00:34.48 | ariel_ | hello all. My ex employer is selling off all there stuff. HEHEHEHE check them out. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3860965320&ssPageName=ADME:B:EF:US:1 |
| 00:34.58 | *** join/#asterisk sivana (~richard@209.91.159.221) |
| 00:34.59 | Moc | it nice, I mean, but very very similar to the IP 500 |
| 00:35.05 | Moc | the display look so nice... |
| 00:35.06 | Brixius | I'm tring to talk to qwest about getting some did #'s sent to me over ip, what do you think the odds of them even finally getting the concept of what I'm asking for down? |
| 00:35.08 | pbd | Now, if I could make CCM call IN to * again.. (that stopped working, I'm guessing codec- haven't dug to hard) |
| 00:35.16 | ariel_ | they are selling there servers, digium cards lots of suff. |
| 00:35.28 | Moc | the text is like all anti-aliase ... very sharp.. |
| 00:35.40 | Moc | the xml browser work well, just got nothing to do with it yet |
| 00:35.59 | Moc | each line have a multi color led associated to it |
| 00:36.12 | jas_williams | Umaro: Post your sip.conf and let me see if it is anything else ? |
| 00:36.54 | AgiNamu | I'm trying to use DDD |
| 00:37.09 | Moc | except that, it mostly the same |
| 00:37.16 | Moc | 6 line.. |
| 00:38.12 | Moc | I guess if you dont need xml and the 6 line, the IP 500 is perfect |
| 00:38.28 | Moc | I hate that it doesnt have a PoE adaptor with it .. |
| 00:38.35 | Moc | I like that of the IP 500 |
| 00:38.41 | AgiNamu | by default, the -g option is used, right? |
| 00:38.46 | AgiNamu | (when compiling asterisk) |
| 00:39.39 | PatrickDK | real poe? or just an injector? |
| 00:40.02 | Moc | ariel_ nice ... |
| 00:40.18 | Moc | ip 600 have the 'real' poe |
| 00:40.33 | Moc | ip 500 need a special cable for 'real' poe.. |
| 00:40.39 | PatrickDK | ya, I can only use real poe, I use all poe enabled switchs |
| 00:40.44 | Moc | but I like the IP 500 power injector |
| 00:40.53 | Moc | PatrickDK, IP 600 then .. |
| 00:41.15 | Moc | alot of stuff there |
| 00:41.21 | Moc | ariel_, nice lot |
| 00:41.25 | PatrickDK | just use a poe injector than |
| 00:41.39 | ariel_ | well it was what were using to develop the asterisk systems |
| 00:41.55 | ariel_ | there are at least 2 digium cards in the servers. |
| 00:42.07 | PatrickDK | the nice thing about 802.3af is that it calculates cable length and compensates for it in the power level |
| 00:42.28 | Moc | I just saw that |
| 00:42.36 | AgiNamu | So I modified channels/Makefile and added -g to the lines with chan_sip |
| 00:42.45 | AgiNamu | that should allow me to use DDD and put a breakpoint in there eh? |
| 00:42.49 | Moc | at 1500 total it cheap.. but Im in canada ;) |
| 00:43.05 | PatrickDK | moc, that is still 3 days away |
| 00:43.13 | Moc | yea |
| 00:43.17 | PatrickDK | I bet it will be atleast triple that cost |
| 00:43.24 | Umaro | http://pastebin.ca/3083 |
| 00:43.28 | Moc | even if it tripple that cost, it still cheap |
| 00:44.03 | Moc | your talking about 2 4 port T1 card + Adtran channel bank |
| 00:44.37 | Moc | + ip phone, + ip ATA |
| 00:44.37 | Umaro | Moc, that sip.conf look ok to you? |
| 00:45.19 | Moc | well I never define my canreinvite global thought.. but yes |
| 00:45.38 | Umaro | hmm |
| 00:45.56 | PatrickDK | hmm, you can't define canreinvite global |
| 00:46.02 | bkw_ | yes you can |
| 00:46.02 | PatrickDK | last time I read chan_sip.c |
| 00:46.07 | bkw_ | you read WRONG |
| 00:46.11 | PatrickDK | heh, must of changed |
| 00:46.14 | bkw_ | no |
| 00:46.14 | ariel_ | Moc, it's going to go over 3k due to I know people that are waiting to place there bids. |
| 00:46.17 | bkw_ | its always been like that |
| 00:46.23 | bkw_ | BZZZT!!! thanks for playing |
| 00:46.25 | Moc | ariel_, for sure ;) |
| 00:46.32 | PatrickDK | heh :) |
| 00:46.38 | PatrickDK | maybe I'm thinking of something else |
| 00:47.01 | bkw_ | static int global_canreinvite = REINVITE_INVITE; |
| 00:47.07 | bkw_ | then it uses that if not set |
| 00:47.12 | Moc | they should make bid that just tell, this Bid will end within the next day. |
| 00:47.21 | bkw_ | I think you're talkin out your ass.. but thats ok |
| 00:47.22 | Moc | so you dont know exactly when that day it will finish .. |
| 00:47.28 | bkw_ | :P |
| 00:47.33 | bkw_ | PatrickDK just giving you a hard time |
| 00:47.35 | bkw_ | its been a long day |
| 00:47.41 | AgiNamu | Can anyone help me setup DDD and Asterisk? |
| 00:47.46 | PatrickDK | I know, heh, been a hell of a day |
| 00:47.56 | AgiNamu | I'd like to debug asterisk , but it wont stop on my breakpoints |
| 00:47.57 | PatrickDK | but then, I went to bed at 3am, and got up at 6 |
| 00:48.32 | bkw_ | at the next astricon... we will have a round of "Who knows the code best!" |
| 00:48.45 | bkw_ | and ask questions about key API's and where they are.. what they do.. and shit |
| 00:48.54 | bkw_ | :P |
| 00:49.04 | PatrickDK | na, lets just have a contest |
| 00:49.19 | bkw_ | um thats what that is :P |
| 00:49.20 | PatrickDK | who can rewrite an api, to make it better :) |
| 00:49.26 | junk-u | bkw_: ya or anthm gonna win? :P |
| 00:49.29 | Umaro | bkw_: can you see anything wrong with this sip.conf? http://pastebin.ca/3083 |
| 00:50.31 | bkw_ | 013 callwaitingcallerid=yes |
| 00:50.31 | bkw_ | 014 threewaycalling=yes |
| 00:50.31 | bkw_ | 015 transfer=yes |
| 00:50.31 | Umaro | it keeps giving me a bad file descriptor message when * starts up and tries to register to a sip peer |
| 00:50.31 | bkw_ | are those EVEN valid in sip.conf |
| 00:50.31 | bkw_ | I think not |
| 00:50.31 | bkw_ | I could be wrong |
| 00:50.31 | bkw_ | but I don't recall those being in there |
| 00:50.31 | Umaro | fair enough |
| 00:50.32 | bkw_ | Umaro describe your problem |
| 00:50.33 | PatrickDK | na, they aren't used in sip.conf |
| 00:50.41 | PatrickDK | it should just ignore them though, I think |
| 00:50.43 | bkw_ | maybe I can tell you what it is |
| 00:50.46 | bkw_ | PatrickDK yep |
| 00:51.02 | Umaro | bkw_: Okay, I need * to do sip on all available interfaces |
| 00:51.04 | bkw_ | we are gearing up for an ICD test sometime soon |
| 00:51.20 | bkw_ | Umaro it does with bindaddr= |
| 00:51.25 | bkw_ | but you need that patch on bugs.digium.com |
| 00:51.30 | bkw_ | so it replies with the correct IP |
| 00:51.46 | Umaro | oh rilly |
| 00:51.51 | Umaro | remember the bug #? |
| 00:51.53 | PatrickDK | yep :) |
| 00:52.08 | PatrickDK | reson why I have sip on internal, and iax on external :) |
| 00:52.33 | *** join/#asterisk lilneon (~tj_r3@cuscon10663.tstt.net.tt) |
| 00:52.37 | lilneon | hi guys |
| 00:53.03 | Umaro | PatrickDK: I tried that, but I get one way audio through IAX for some odd reason |
| 00:53.08 | lilneon | hey has anyone gotten festival to work? |
| 00:53.58 | lilneon | hello? |
| 00:54.05 | Moc | hey |
| 00:54.11 | lilneon | hi |
| 00:54.12 | PatrickDK | umaro, odd, I never had that problem with iax |
| 00:54.31 | lilneon | Moc: ever got festival to wok with asterisk? |
| 00:54.35 | PatrickDK | asterisk isn't behind some kind of firewall is it? |
| 00:54.36 | Moc | sure |
| 00:54.40 | Chuji | lilneon : "Work" is a sort of an oxymoron with festival |
| 00:54.49 | lilneon | chuji: why? |
| 00:54.54 | Moc | hehe |
| 00:54.57 | Chuji | can be flakey |
| 00:55.02 | Chuji | but it will "work" |
| 00:55.09 | Moc | lilneon, www.voip-info.org is your best friend |
| 00:55.12 | Chuji | festival's voices are ass |
| 00:55.14 | lilneon | Moc: did u have to download a patch or get it running with it installed via the linux rpm? |
| 00:55.26 | lilneon | moc: tried that |
| 00:55.27 | Chuji | dont use the rpm |
| 00:55.39 | Umaro | PatrickDK: nope. really strange. actually though, I think it might be a problem with my firefly iax clients.. trying to switch them over to sip, and that seems to be working ok |
| 00:55.40 | Moc | dont recall, I dont use it anymore |
| 00:55.53 | Moc | I got some other one that cost 30$, alittle better |
| 00:56.04 | lilneon | Moc: name? |
| 00:56.14 | lilneon | maybe i could follow in your fotsteps |
| 00:56.18 | Moc | lilneon: http://www.cepstral.com/ |
| 00:56.28 | Chuji | That is what I use too |
| 00:56.34 | Chuji | but I don't use it streaming |
| 00:56.34 | lilneon | thnx Mox |
| 00:56.38 | lilneon | moc* |
| 00:56.47 | lilneon | chuji: any specific reason? |
| 00:56.51 | Chuji | I just create the .wav dynamicall, then background it |
| 00:57.29 | Chuji | lilneon : I had better results that way |
| 00:57.33 | Chuji | slow box |
| 00:57.47 | Chuji | I can use nice to prioritize it |
| 00:57.52 | Moc | bkw_ no more MoH ? |
| 00:58.18 | lilneon | chuji: hmm ok.. |
| 00:58.20 | ccfiel | can somebody help me... after 10 mins my sip registration disconnect.. is there any thing i did wrong? |
| 00:58.57 | lilneon | chuji: so wat did u use? AGI to create the wav? |
| 00:59.33 | Chuji | lilneon : Yeah, it's in my ivr applications |
| 00:59.49 | Chuji | lilneon : I have one in my dialplan that just uses exec |
| 01:00.09 | Chuji | lilneon: It creates a file based on the CIDname then calls my cell and announces it |
| 01:01.01 | Slimey | woohoo, chan_sccp works on solaris to my 7902 |
| 01:01.10 | lilneon | chuji:nice man nice.. .. but u clean up after wards right? cuz i can see that dumping alot of.wav files on my poor lil box |
| 01:01.11 | Chuji | Thanks to anthm's pimp-a-licious macro patch in app_dial, that functionality works real well |
| 01:01.32 | Chuji | lilneon: Of course |
| 01:01.47 | lilneon | chuji: just making sure.. i think i will do the same |
| 01:01.56 | lilneon | thnx |
| 01:02.21 | Chuji | lilneon : You can test the voice, it just puts some bs in it like "to register this voice go to www.blah blah" |
| 01:03.15 | lilneon | chuji:ok |
| 01:03.22 | Chuji | Now if you want to play with a KICK ASS voice |
| 01:03.24 | Chuji | http://www.rhetorical.com/cgi-bin/demo.cgi |
| 01:03.32 | Chuji | check out that valley girl voice |
| 01:04.43 | Chuji | I like having her tell me she wants to lick the underside of my nutsack. Cracks me up every time |
| 01:05.00 | modulus_ | omg rhetorical is hot |
| 01:05.01 | lilneon | chuji:.. lol.. yeah she sounds real seductive |
| 01:06.01 | Chuji | Yeah, thats the best voice I've found yet |
| 01:06.13 | Chuji | wish they would gpl that suckah |
| 01:07.34 | tzanger | whew |
| 01:07.59 | tzanger | I thought I fucked up my Norstar system, it wasn't getting any calls to our main business line. I just had a password wrong in asterisk though :-) |
| 01:08.11 | modulus_ | she sounds nothing like a valley girl though |
| 01:08.21 | Chuji | modulus_ : You don't think? |
| 01:08.25 | modulus_ | nope |
| 01:08.32 | Chuji | modulus_ : You chose valley girl right? |
| 01:08.41 | modulus_ | yup |
| 01:08.45 | *** join/#asterisk _chad (~Chad@c-24-6-142-55.client.comcast.net) |
| 01:08.49 | Chuji | hmm, I thought it was pretty good |
| 01:09.00 | Atacomm | you guys see the new Gigabyte graphics card with dual GeForce 6600 chips on it? !!! wicked.... |
| 01:09.20 | Chuji | does it make xterm better? |
| 01:09.31 | puzzled | vi go faster? |
| 01:10.01 | modulus_ | maybe it makes xscreensaver render more smoothly |
| 01:10.08 | Atacomm | hehe |
| 01:10.27 | Atacomm | i wonder if you can use two in a system :) |
| 01:10.32 | puzzled | hehe, opengl ones are way too buggy to render anything without crashing the entire box |
| 01:10.52 | PatrickDK | I personally perfer the new matrox quad cards |
| 01:11.39 | _chad | anyone else grab that cisco79(46)0 torrent before it died? |
| 01:11.49 | _chad | or am i hosed |
| 01:11.54 | modulus_ | you're hosed |
| 01:11.56 | Atacomm | i've got MPC sending me a Xeon box with 5v pci, 3v pci-x and two pci-express slots for testing our cards with :) |
| 01:12.01 | *** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
| 01:12.08 | Atacomm | talk about a powerful and flexible platform |
| 01:12.12 | _chad | sucks to be hosed |
| 01:13.04 | modulus_ | rhetorical is a little better that at&t's text to speech |
| 01:13.39 | JerJer | is hosed better or worse then corncobbed? |
| 01:13.47 | modulus_ | hi jerjer |
| 01:14.44 | *** join/#asterisk arbius (~arbius@c-67-173-45-26.client.comcast.net) |
| 01:14.44 | modulus_ | it can't be worse than being hot-carled |
| 01:14.50 | puzzled | JerJer: do you really like the P104's? |
| 01:14.56 | JerJer | yeah they rock |
| 01:15.13 | JerJer | ye of little faith |
| 01:15.33 | puzzled | JerJer: amazing. the P103 was a piece of crap |
| 01:15.35 | AgiNamu | so, can someone help point me in the right direction in as far as debugging chan_sip |
| 01:15.42 | puzzled | guess the learnt from their mistakes |
| 01:15.42 | AgiNamu | I could just post to the bug tracker and forget about it |
| 01:15.58 | AgiNamu | but A: I wanna fix it asap, cause it's a showstopper at this point and B: I'd like to learn |
| 01:16.02 | *** join/#asterisk seong (~seong@203.115.208.140) |
| 01:16.31 | *** join/#asterisk bjohnson (~bjohnson@ip138-172.dsl.istop.com) |
| 01:16.39 | *** join/#asterisk iris___ (iris@coredump.servercity.com) |
| 01:16.46 | iris___ | evening all |
| 01:17.25 | iris___ | was wondering if someone might be ablt to answer some questions about how to setup g726? |
| 01:17.45 | puzzled | iris___: buy the licenses, activate the codec |
| 01:18.09 | D1ng0 | question asterisk behind firewall, remote client connecting through redirected ports thru firewall, can auth okay to * and call out but no sound... message udp port 8000 unreachable... what gives |
| 01:18.09 | puzzled | speaking of "worse": http://laimbock.com/tmp/worse.jpg |
| 01:18.25 | iris___ | i was told g726 was license free, and that g729a is licensed? |
| 01:18.43 | puzzled | iris___: oops I read g729. sorry |
| 01:18.48 | iris___ | no prob ;) |
| 01:19.10 | iris___ | i kee getting errors when i put allow=g726 or allow=g72632k in my iax.conf |
| 01:19.11 | Chuji | puzzled must not know his 6's from his 9's, so his girlfriend says ;) |
| 01:19.29 | puzzled | lol |
| 01:19.33 | AgiNamu | is sip_pvt.fromuser used instead of .from when setting the From: header? |
| 01:20.08 | Atacomm | anyone ever used a carrier access channel bank I? i'm looking at a used unit and havent found it on the CA website... |
| 01:20.11 | Chuji | iris___ : IIRC * only supports 32k |
| 01:20.45 | iris___ | then shouldn't "allow=g72632k" work? or do i have the wrong synyax? |
| 01:21.02 | Chuji | I'd go with g726 |
| 01:21.10 | Chuji | grep the iax code to be sure |
| 01:21.11 | *** join/#asterisk Ya_Ba (~YaBa@82.155.36.239) |
| 01:21.54 | iris___ | allow=g726 also give errors ;( |
| 01:21.57 | iris___ | let me try again. |
| 01:22.08 | Chuji | sure you got the order right? |
| 01:22.12 | Chuji | disallow all first |
| 01:22.27 | iris___ | yes disallow=all |
| 01:22.29 | iris___ | first |
| 01:22.34 | Chuji | groovy |
| 01:23.51 | iris___ | Dec 16 20:32:25 NOTICE[30624]: chan_iax2.c:5730 socket_read: Rejected connect attempt from 66.234.228.170, requested/capability 0x4/0x42e incompatible with our capability 0xf810. |
| 01:23.58 | iris___ | using allow=g726 |
| 01:24.58 | *** join/#asterisk badc{}l{}r (~badc{}l{}@203.115.208.140) |
| 01:25.05 | Chuji | iax2 debug |
| 01:25.12 | Chuji | make sure it's requesting 726 |
| 01:27.19 | iris___ | i dont see which codec it is requesting but i do see " CAUSE : Unable to negotiate codec |
| 01:27.25 | iris___ | just after the RSA result |
| 01:27.42 | Chuji | hmm, I thought it was in the setup |
| 01:27.50 | Chuji | Is this to a voip provider? |
| 01:27.58 | Chuji | Or peer to peer |
| 01:28.05 | iris___ | yes incoming call from voicepulse |
| 01:28.37 | Ya_Ba | anyone running asterisk + Festival ? it doesn't work here :( |
| 01:28.40 | ccfiel | can somebody help me... after 10 mins my sip registration disconnect.. is there any thing i did wrong? |
| 01:28.44 | iris___ | i spoke with them today about poor incoming call quality (choppy etc) and they said to try g726 |
| 01:29.01 | ccfiel | can somebody help me... after 10 mins my sip registration disconnect.. is there any thing i did wrong? |
| 01:29.08 | Chuji | Ya_Ba : Uhhg we just went over that |
| 01:29.20 | ccfiel | sorry for the doube post..:( |
| 01:29.25 | Ya_Ba | missed that :P |
| 01:29.33 | D1ng0 | ccfiel, possible lag ??? |
| 01:30.11 | ccfiel | D1ng0: i thinks its not..because its in the same network.. |
| 01:30.24 | D1ng0 | whats sip show peers say ? |
| 01:30.49 | ccfiel | D1ng0: phone1/phone1 192.168.2.108 D 255.255.255.255 5060 Unmonitored |
| 01:30.49 | ccfiel | 1 sip peers loaded [1 online , 0 offline] |
| 01:31.00 | AgiNamu | no one here familiar with chan_sip.c ? |
| 01:31.08 | Ya_Ba | Chuji, i'm using XorCom ... debian+asterisk, it cames already with Festival, I edited the festival.conf file, but still doesn't work :| |
| 01:31.31 | junk-u | ccfiel: add qualify=yes in ur sip.conf for that peer |
| 01:31.51 | iris___ | Chuji: any other ideas on g726? |
| 01:32.00 | ccfiel | junk-u: ok ill try..:) |
| 01:32.37 | Chuji | Ya_Ba : cuddle up with the wiki a bit, there's some festival stuff there |
| 01:32.44 | junk-u | do a reload and sip show peers again |
| 01:32.52 | junk-u | unmonitored should be changed. |
| 01:33.10 | Chuji | iris___ : It really sounds like they aren't allowing 726 comign to you |
| 01:33.21 | Chuji | iris___ : Throw your logger.conf in debug too |
| 01:33.31 | Chuji | iris___ : See if you get any additional help there |
| 01:33.51 | iris___ | Chuji: will do, thanks!! |
| 01:34.04 | Ya_Ba | Chuji, i'm there... it just talks about the .conf file, done that alread |
| 01:34.04 | Chuji | iris___ : Short of that, you might want to tcpdump it, and grep for 726 |
| 01:34.12 | RestLessGemini | ccfiel: also check maxexpirey and defaultexpirey in general section of sip.conf, try playing with both values |
| 01:34.20 | ccfiel | junk-u: my show peers show like this LE |
| 01:34.20 | ccfiel | 1 sip peers loaded [0 online , 1 offline] |
| 01:34.20 | ccfiel | localhost*CLI> ò |
| 01:34.27 | D1ng0 | okay where in the wiki is informationj on connecting asterisk office 1 to asterisk office 2 |
| 01:34.51 | Chuji | Ya_Ba : there is some patching that needs to be done, iirc |
| 01:34.51 | ccfiel | junk-u: unreachable is the status now.. |
| 01:34.55 | ariel_ | D1ng0, look for iax2 trunking. |
| 01:35.02 | D1ng0 | ariel_, cool |
| 01:35.04 | junk-u | cause that peer isnt online. |
| 01:35.25 | AgiNamu | Anyone else here using a proxy for load balancing? |
| 01:35.29 | RestLessGemini | ccfiel: you can put qualify=[value in ms] |
| 01:35.29 | AgiNamu | in front of asterisk |
| 01:35.54 | Chuji | D1ng0 : Also, you can share dialplans with iax2. That's in addition to trunking |
| 01:36.03 | Chuji | again, can be found on the wiki |
| 01:36.47 | ariel_ | AgiNamu, last setup I did they put 2 tnt switches between the 9 asterisk servers. |
| 01:40.39 | Moc | ok... I 3 IP phone in front of me... 1 IP 500, 1 IP 600 and 1 7960... let write a comparaison of each |
| 01:41.31 | Chuji | leaving price out of the picture? |
| 01:42.06 | Moc | I'll try to put everything I can think off |
| 01:42.48 | Moc | 1 strange thing... I get less lag with the polycom than the cisco !!! |
| 01:43.06 | AgiNamu | so.... last call :( -- no one here familiar with chan_sip.c? |
| 01:45.35 | Moc | someone can go on bkw conference, I wanna ask what phone sound the best in your opinions ? |
| 01:45.53 | junk-u | moc: wait |
| 01:47.51 | Moc | the handfree of both polycom ip 500 and 600 sound better |
| 01:48.25 | PatrickDK | hmm, raid5's transfer rates suck |
| 01:48.33 | ccfiel | RestLessGemini : i have tried to put 999999 but the status says unreachble |
| 01:49.23 | *** join/#asterisk pchenry (~pchenry@firewall1.enetonline.net) |
| 01:49.27 | D1ng0 | bahhhh No registration for peer 'remote' |
| 01:49.33 | pchenry | Wassup guys |
| 01:49.36 | Moc | hi |
| 01:49.40 | *** join/#asterisk tessier (~treed@ip-66-181-5-46.cust.i2bnetworks.com) |
| 01:49.41 | tessier | Hello all |
| 01:49.46 | Moc | hi tessier |
| 01:49.58 | D1ng0 | following the wiki it doesnt work |
| 01:50.13 | pchenry | Someone wanna drop some opinions about how to setup SIP extensions in call groups without using an AGI |
| 01:50.29 | Chuji | app_queue? |
| 01:50.49 | pchenry | Chuji ... was that for me ? |
| 01:50.52 | Chuji | yeah |
| 01:51.02 | Chuji | unless I'm misunderstanding what you want to do |
| 01:51.04 | pchenry | I dont wanna pt them in a queue |
| 01:51.14 | pchenry | I wanna ring multiple extensions at the same time |
| 01:51.18 | Chuji | use the '&' in your dial string |
| 01:51.20 | pchenry | SIP extensions |
| 01:51.27 | pchenry | so thats works well |
| 01:51.39 | pchenry | I saw it but seemed to be a nasty way of doing it |
| 01:51.54 | PatrickDK | why nasty? |
| 01:51.56 | Chuji | yeah, pretty nasty if you have a lot of them |
| 01:51.59 | pchenry | there is a callgroups parameter for sip.conf ... at least the WIKI says so |
| 01:52.01 | PatrickDK | how else would ya do it? |
| 01:52.03 | pchenry | but no description |
| 01:52.15 | Chuji | pchenry : that is for pickup I bet |
| 01:52.18 | PatrickDK | you can always use LOCAL/XXXX |
| 01:52.18 | pchenry | ahhh |
| 01:52.19 | Moc | oh well why do I botter... I know polycom is better... |
| 01:52.28 | PatrickDK | and have XXXX dial a group of them |
| 01:52.32 | dan2 | bkw_: ping |
| 01:52.34 | Chuji | Moc: So send me the 7960 |
| 01:52.46 | Moc | Chuji, I'll switch with a polycom ip 500 |
| 01:52.57 | Chuji | Really? |
| 01:53.03 | pchenry | BTW ... I was here asking about upgrading the 7960 from SCCP to the lastest SIP last night |
| 01:53.09 | pchenry | I have a good routine for it now |
| 01:53.21 | implicit | i would trade 7960 for ip 600 if anyone wants |
| 01:53.21 | implicit | ? |
| 01:53.36 | junk-u | ive an ip500 too |
| 01:53.38 | Moc | implicit, yea I should have said a ip 600 too hehe ;) |
| 01:53.40 | junk-u | it rocks! |
| 01:53.43 | D1ng0 | Chuji, bahhh is this iax2 peer document in the wiki right |
| 01:53.54 | implicit | Moc: ip500 feels janky though |
| 01:53.56 | Moc | but I'll keep this cisco to show people the difference. .. |
| 01:53.57 | implicit | compared to 600 |
| 01:53.59 | Chuji | D1ng0 : dunno |
| 01:54.05 | Chuji | D1ng0 : how old is it? |
| 01:54.06 | Moc | implicit, janky ? |
| 01:54.09 | D1ng0 | well shooooot |
| 01:54.09 | implicit | yeah |
| 01:54.19 | dan2 | any ops here? |
| 01:54.24 | Moc | what that mean ? |
| 01:54.40 | dan2 | can anyone verify if the rfc2833 fixes made by bkw have been pushed into stable |
| 01:54.47 | Moc | because both phone are nearly identicle |
| 01:54.55 | pchenry | seem you guys like the polycom phones better than cisco ? |
| 01:54.56 | Moc | but I like better the handfree button |
| 01:55.02 | Moc | pchenry oh yea |
| 01:55.06 | implicit | Moc: the 600 is more solid |
| 01:55.06 | D1ng0 | <PROTECTED> |
| 01:55.07 | Chuji | danf: Check the bug tracker |
| 01:55.12 | Chuji | dan2 rather |
| 01:55.16 | pchenry | implicit ... whatt ya like better ? |
| 01:55.17 | tessier | pchenry: Cisco is a real PITA. :( |
| 01:55.25 | tessier | I understand the polycom phones can do busy lamps like the snoms too. |
| 01:55.28 | dan2 | Chuji: it was a fix, it was never posted to bug tracker iirc |
| 01:55.32 | pbd | tessier: No kidding. |
| 01:55.32 | Moc | implicit, it the exact same casing from what I see |
| 01:55.33 | tessier | I am really looking forward to using only polycom in the future. |
| 01:55.35 | pchenry | tessier ... you were here last night ... I got all of my phones up'd |
| 01:55.39 | tessier | At the moment I am using a mix and it is a nightmare. |
| 01:55.42 | Chuji | dan2 : Then check the changelog |
| 01:55.46 | tessier | pchenry: You go them all flashed? |
| 01:55.48 | pchenry | yes |
| 01:55.56 | pchenry | 2 probs to get passed .... |
| 01:56.08 | tessier | pchenry: Congrats. Took me ages to figure it out. A couple of months. Wish someone would have been here to help me back then. |
| 01:56.12 | Moc | tessier, I love polycom configuration system.. it is so advanced.. |
| 01:56.12 | pchenry | the filename issue and the 4k size limit on SIPDefaults.cnf |
| 01:56.45 | Moc | check the beggining of this page to see some stuff I like about the polycom : http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones |
| 01:56.50 | pchenry | k |
| 01:57.07 | Moc | IP 600 display look so nice.. |
| 01:57.12 | Moc | I gota try uploading font file. |
| 01:57.19 | symlink[laptop] | dan2: yes |
| 01:57.31 | Moc | and I gota try to make a 3l33t language file ;) |
| 01:57.40 | pbd | Anyone out there an h.323 maven? Dealt with prefixes, routing, and gnugk? |
| 01:57.48 | Moc | h0ld, 3nd c4ll |
| 01:58.08 | Chuji | pbd : That would be jerjer |
| 01:58.34 | junk-u | MOC |
| 01:58.37 | junk-u | sing! |
| 01:58.41 | Moc | no way ... ;) |
| 01:58.47 | pbd | I'm guessing he's off to dinner or some such- what I need is probably staring me in the face. |
| 01:58.55 | D1ng0 | heck this wiki has to be wrong.... i think its written backwards |
| 01:58.59 | junk-u | everyone go in the conf and sing happy birthday for my gf! |
| 01:59.02 | Chuji | pbd : Nah, he's always lurking |
| 01:59.12 | Chuji | just call him an asshole or something... He'll chime up |
| 01:59.22 | Moc | ha the handset of the polycom are longer |
| 01:59.22 | Chuji | or say voipjet is better than nufone |
| 01:59.29 | pbd | And it's a &(*($#$ Cisco integration problem, to boot. |
| 01:59.57 | pbd | JerJer- Chuji said it, not me... but if you've now woken up... ;-) |
| 02:00.32 | pbd | Cisco CCM has this annoying configuration policy on H.323/H.225 trunks.. |
| 02:00.45 | pbd | They expect a prefix, but they don't register it with the gatekeeper.. |
| 02:01.06 | pbd | and they don't have a way to strip it off before they do call routing, other than MSD stripping. |
| 02:01.55 | pbd | Problem is- I've got 4 digit extensions, *AND* I want to use the CCM to route outbound PSTN calls. |
| 02:02.17 | pbd | I think I've got an ugly answer- that's to set up two different trunks and prefixes under CCM. |
| 02:02.38 | pbd | But I'm hoping I can set up a more general way, where I can get gnugk to simply pass all digits on to the callmanager. |
| 02:02.46 | pbd | *WITHOUT* a prefix. |
| 02:03.41 | pbd | My ugly solution also doesn't account for some dial plans on the CCM side that actually use a variable number of digits- it would mean a LOT of trunks. |
| 02:04.34 | pbd | (last count, something like 87 trunks to register with the gatekeeper- internal corporate toll bypass system, not IP based) |
| 02:04.55 | pbd | Anybody out there have a way to rig that one? |
| 02:05.32 | tessier | I don't care how much they offer to pay me. |
| 02:05.35 | tessier | It is always a money loser. |
| 02:05.39 | Nugget | heh |
| 02:05.51 | pbd | Under CCM 3.3, it's the ONLY way to fly. Literally. |
| 02:06.09 | Nugget | I want an h323 solution in perl written agains mysql in linux 2.6 running amd64 native binaries. :) |
| 02:06.11 | *** join/#asterisk Umaro (~umaro@c-24-22-76-14.client.comcast.net) |
| 02:06.25 | Nugget | let's see, how can I make that even worse... |
| 02:06.31 | pbd | CCM4 gives me SIP trunks- but the timeline for us doing that is still a few months off. |
| 02:06.34 | Umaro | Ok guys, the next astricon needs to be in portland. My reasoning: http://www.markspencer.com |
| 02:06.43 | badc{}l{}r | how many call forwarding supported by *? |
| 02:07.17 | Chuji | Umaro : Heh, don't think they would host a "con" |
| 02:07.22 | junk-u | Umaro: mouhahahha |
| 02:07.26 | badc{}l{}r | sorry, rephase .. how many call forwarding functions supported by * |
| 02:07.35 | Nugget | hahah Umaro. |
| 02:07.37 | Chuji | badc{}l{}r : As in? |
| 02:08.00 | Chuji | badc{}l{}r : like CFA CFBN CFNA ? |
| 02:08.10 | Umaro | also, because i'm in portland, and then I wouldn't need to worry about airfare ;) |
| 02:08.52 | Moc | ok bkw conference sound bad ... |
| 02:09.38 | Moc | g726 ... urm |
| 02:09.50 | badc{}l{}r | CFNA |
| 02:10.00 | badc{}l{}r | and CFBN |
| 02:10.09 | D1ng0 | closer, okay register_verify: Peer 'remote' is not dynamic (from 67.8.205.130) |
| 02:10.35 | tessier | Can an extension be a channel somehow? |
| 02:10.36 | PatrickDK | d1ng0, don't register static hosts |
| 02:10.53 | tessier | a MeetMe conference that is |
| 02:10.57 | Moc | 1 thing that aint bad on the cisco 7960, is they put a pause when you press the mute or speaker button, so you dont hear pressing it |
| 02:11.11 | tessier | At a certain time I want to put an extension into the conference that plays a beep or something. |
| 02:11.16 | Chuji | badc{}l{}r : Well, that's kind of a trick question. It's supported by the technology, i.e. sip, zap, etc, and.. it's supported in the dialplan logic. But you will need to roll your own there |
| 02:11.38 | twisted | moc: it doesn't make any noise, how would you hear it in the first place? |
| 02:12.12 | D1ng0 | PatrickDK, then how do they know to connect ? |
| 02:12.16 | Moc | you hear it on the polycom |
| 02:12.31 | PatrickDK | d1ng0, cause you made it static |
| 02:12.37 | badc{}l{}r | thanks citats |
| 02:13.05 | Moc | i saw the difference, once you press the button on the polycom, the sound is automaticly put back, the cisco has a delay. It could be only the phone being slow too |
| 02:13.08 | PatrickDK | when it's a database |
| 02:14.12 | twisted | Moc, the cisco has no delay |
| 02:14.36 | Moc | it small but it there |
| 02:14.50 | D1ng0 | PatrickDK, no without a register line it never tried to connect to the remote |
| 02:14.56 | twisted | if small == <10ms then yea |
| 02:14.59 | D1ng0 | hrmmm |
| 02:15.14 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 02:15.14 | PatrickDK | d1ng0, it's not suppost to, unless you use a dial command |
| 02:15.28 | Moc | that is why cisco audio is always alittle lagged too |
| 02:15.45 | PatrickDK | moc, what codec you using on the cisco? |
| 02:15.50 | Moc | ulaw |
| 02:15.55 | twisted | moc, whatever man... i'm getting fed up with your cisco bashing |
| 02:16.29 | paulc | I use my Cisco 7940 daily and haven't really got any problems with it |
| 02:16.35 | paulc | each to their own, no? |
| 02:16.40 | twisted | i have never had problems out of the 7960's i've used either |
| 02:16.42 | Moc | twisted, im not bashing at cisco, I find it pretty good to not hear when pressing the button... |
| 02:16.48 | *** join/#asterisk sumasuma (~suma@81-86-240-118.dsl.pipex.com) |
| 02:16.51 | twisted | not hear what?! |
| 02:16.53 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 02:16.58 | twisted | there's nothing to hear, the button makes NO NOISE |
| 02:17.02 | paulc | DTMF issues? |
| 02:17.07 | sumasuma | may i know which PCI Radio card will work with asterisk ? |
| 02:17.13 | twisted | paulc, no, speaker or mute |
| 02:17.26 | Moc | twisted, just touch your speaker or mute button, dont press it, and you will hear it on the other end.. |
| 02:17.36 | sumasuma | i want to put FM Radio as musiconhold |
| 02:17.38 | twisted | let me test |
| 02:17.39 | Moc | but if you press to mute or speaker phone, you wont hear it |
| 02:18.04 | paulc | just touching the button? cos the microphone is positioned right below those buttons? |
| 02:18.08 | D1ng0 | PatrickDK, so if its not dynamic, the client server will not connect to the primary unless someones trying to dial ? |
| 02:18.20 | PatrickDK | yep |
| 02:18.40 | twisted | moc, i called my other phone from the cisco |
| 02:18.45 | twisted | i touched the mute/speaker button |
| 02:18.46 | D1ng0 | okay but if the client is dynamic, it should register |
| 02:18.47 | twisted | heard nothing |
| 02:18.59 | sumasuma | D1ng0: May i know which PCI Radio card will work with asterisk ? I want to put FM Radio as musiconhold |
| 02:19.02 | PatrickDK | yep |
| 02:19.02 | twisted | the only way I could hear it is if i put the phone on speaker and flicked the buttons |
| 02:19.16 | D1ng0 | sumasuma, no idea, over my head |
| 02:21.52 | twisted | I spy with my little eye something beginning with ([a-zA-Z0-9].*) |
| 02:22.50 | D1ng0 | PatrickDK, okay im lost read the wiki, if ya ask me its way wrong, the other remote server keeps saying... register_verify: Peer 'remote' is not dynamic when it is... what did i miss |
| 02:22.58 | *** part/#asterisk kongnamool (~sexton@67.113.12.162) |
| 02:23.05 | pbd | Wow, that's cool.. JerJer, now I can call you an ass. :) |
| 02:23.13 | pbd | Set up my ugly solution.. |
| 02:23.15 | PatrickDK | d1ng0, post the iax.conf |
| 02:23.17 | pbd | Works.. kinda.. |
| 02:23.36 | D1ng0 | PatrickDK, from both client and remote to pastbin.ca |
| 02:23.39 | pbd | For 4 digit dialing, which goes to Cisco 7960 on my desk- works swimmingly. |
| 02:23.56 | pbd | For PSTN termination.. rings the phone- and then dumps the call 2 seconds after answer. |
| 02:23.56 | PatrickDK | yep |
| 02:24.12 | PatrickDK | pbd, I had that problem |
| 02:24.20 | PatrickDK | the fix is to do something, hmm |
| 02:24.33 | pbd | Patrick: Now, the What is the interesting question, isn't it? |
| 02:24.38 | PatrickDK | been along time, forget |
| 02:24.59 | pbd | This didn't work at all until today, latest head. |
| 02:25.28 | PatrickDK | no, I had the call drop 2 seconds after answer, well, used too, like half a year ago |
| 02:25.39 | snewpy | pbd: h323? |
| 02:25.39 | PatrickDK | connecting my cisco stuff to asterisk |
| 02:25.48 | pbd | Snewpy- yep. |
| 02:26.00 | D1ng0 | PatrickDK, http://pastebin.ca/3095 |
| 02:26.07 | junk-u | pdb: asterisk-cvs-request@lists.digium.com ? |
| 02:27.43 | *** join/#asterisk funkknob (~funkknob@203.223.34.226) |
| 02:27.46 | *** join/#asterisk Darwin35 (~darwin35@c-24-3-204-71.client.comcast.net) |
| 02:27.48 | PatrickDK | d1ng0, heh, the client should not register with the primary, cause in primay host=(ip address) |
| 02:27.48 | pbd | junk-u ?? |
| 02:27.50 | *** join/#asterisk SexyKen (~sexyken@64-121-7-74.c3-0.snmt-ubr2.sfrn-snmt.ca.cable.rcn.com) |
| 02:28.02 | PatrickDK | but primary should regiser with client, cause in client host=synamic |
| 02:28.05 | pbd | Patrick- were you ever successful? |
| 02:28.11 | SexyKen | Hey guys is it normal that a conference call can only hold 4 people onit? |
| 02:28.12 | pbd | In connecting CCM to *? |
| 02:28.16 | PatrickDK | pbd, yep |
| 02:28.26 | PatrickDK | I didn't do ccm to * |
| 02:28.30 | snewpy | pbd: it's an RTP problem if it gets cleared after the CONNECT |
| 02:28.32 | junk-u | the mailing list asterisk-cvs-request@lists.digium.com gives all the modifs in cvs. |
| 02:28.35 | pbd | Ken- MeetMe, I've had over 25 on at a time. |
| 02:28.35 | PatrickDK | did cisco 1760 t1 to asterisk |
| 02:28.43 | snewpy | pbd: what cause code is CCM returning in the RELEASE COMPLETE msg? |
| 02:28.51 | SexyKen | •pbd• That's wierd, I only have 4 at a time for some reason |
| 02:28.52 | pbd | Snewpy- so... codec mismatch? |
| 02:29.08 | D1ng0 | PatrickDK, HAH i was write the wiki is backwards then, i should simply reverse iax.confs |
| 02:29.17 | snewpy | pbd: probably not codec mismatch, more likely a packetization mismatch, or payload type mismatch or something |
| 02:29.30 | SexyKen | •pbd• Do I have to open more channels or something? |
| 02:29.43 | PatrickDK | d1ng0, well, I think you just want to swap the remote sections |
| 02:29.49 | pbd | snewpy- hangon, I'll check it out. |
| 02:29.53 | PatrickDK | to make the client register to the primary |
| 02:30.12 | PatrickDK | if the client ip changes, but the primary stays consistent |
| 02:30.32 | PatrickDK | well, just swap host lines really |
| 02:30.41 | Darwin35 | [pbx_loopback.so]Dec 16 21:30:21 WARNING[647]: loader.c:258 ast_load_resource:/usr/lib/asterisk/modules/pbx_loopback.so: Undefined symbol "pbx_substitute_variables_varshead" |
| 02:30.41 | Darwin35 | Dec 16 21:30:21 WARNING[647]: loader.c:440 load_modules: Loading module pbx_loopback.so failed! |
| 02:30.41 | Darwin35 | Ouch ... error while writing audio data: : Broken pipe |
| 02:30.41 | Darwin35 | mypbx# |
| 02:30.43 | pbd | Cause 7, resource unavailable. |
| 02:31.01 | pbd | Ken- I didn't have to mess with it, much.. |
| 02:31.09 | pbd | how are you connecting to the meetme? |
| 02:31.13 | pbd | sip? h.323? |
| 02:31.38 | snewpy | pbd: can you send me an ethereal trace of the call? |
| 02:31.39 | SexyKen | I have an SIP phone connecting to the network directly then I have an iax did |
| 02:31.53 | D1ng0 | PatrickDK, got it... its working now... someone needs ta fix the wiki |
| 02:31.57 | Darwin35 | ? |
| 02:32.17 | pbd | Snewpy- yes, but it might take me a few to set up. You want the tcpdump packets, or do you want me to dump out the ethereal? |
| 02:32.26 | D1ng0 | hehehe now to tackle the dial-plan and extensions :) |
| 02:33.26 | Darwin35 | anyone having this problem |
| 02:33.40 | pbd | Snewpy- I'm really guessing codecs here, since cisco lets me specify different ones for different routing patterns.. I haven't dug into it yet, though. I've restricted my h.323 to be ulaw only (cisco doesn't like gsm much) |
| 02:34.18 | snewpy | pbd: as a pcap file including both port 1720, and the rtp traffic |
| 02:34.40 | Darwin35 | ? |
| 02:35.34 | pbd | Snewpy- ok, give me a little time. |
| 02:35.45 | snewpy | pbd: ok |
| 02:36.32 | Darwin35 | http://pastebin.ca/3096 |
| 02:36.43 | Darwin35 | can some one help with this |
| 02:41.38 | pchenry | whats everyones opinion of AMP |
| 02:41.52 | Moc | AMP can be dangerous ... ;) |
| 02:42.50 | pbd | Snewpy- ok, definite squirrelyness going on here. I did a little more testing before I snarf the packets- I've got one-way audio as well. * to CCM works fine.. CCM to * is silent, using DIAX. |
| 02:43.30 | pbd | DIAX to * termination I get two way audio, so it would appear to be an RTP issue. |
| 02:43.30 | Darwin35 | grrr |
| 02:43.54 | snewpy | pbd: ok, send me a trace in both directions |
| 02:47.39 | tessier | Dec 16 18:46:28 WARNING[3492]: app_queue.c:322 changethread: Can't change device with no technology! |
| 02:47.42 | tessier | Now what the heck does this mean? |
| 02:48.05 | Moc | anyone here played with a 7960 or 7940 on a Cisco Call Manager, and on SIP ? |
| 02:48.31 | |Vulture| | Moc: you get your 600 yet? |
| 02:48.41 | Moc | |Vulture|, yeaaaa |
| 02:48.47 | |Vulture| | how is it next to the 500s? |
| 02:48.55 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 02:49.37 | Darwin35 | gc |
| 02:50.01 | Moc | taller lol.. well mostly the same ... except button is place different, the speaker, handset and mute button have a led like the 7960 |
| 02:50.38 | tessier | wtf? This used to work! |
| 02:50.38 | Moc | and 1 nice feature, each line have a multi color LED, so you can see the status of the line more easily |
| 02:50.52 | *** part/#asterisk lilneon (~tj_r3@cuscon10663.tstt.net.tt) |
| 02:51.12 | Moc | the display is higher res, and really look asome... you can't see the pixel (unless you are really close) |
| 02:51.19 | Moc | but that about it |
| 02:51.28 | tessier | Why do I keep getting hit with the problems that someone asks once on a mailing list and nobody replies with a solution? |
| 02:51.40 | tessier | Just one hit on this error message other than the asterisk source and nobody replied. |
| 02:52.21 | Moc | the xml browser work, and is very nice... |
| 02:54.08 | Darwin35 | anyone ? |
| 02:54.14 | Darwin35 | I need my * back |
| 02:54.54 | symlink[laptop] | Darwin35: rm -rf /usr/lib/asterisk/modules |
| 02:54.58 | symlink[laptop] | then do a make install in the asterisk directory |
| 02:55.08 | Moc | you can't set more than the name and phone number on the 7960 ? or there is a option to unlock more field ? |
| 02:55.19 | dan2 | bkw_: wake up |
| 02:55.35 | symlink[laptop] | dan2: are you still asking the question? |
| 02:55.47 | dan2 | symlink[laptop]: rfc2833 is still broke |
| 02:56.25 | symlink[laptop] | so try, like, info! |
| 02:56.37 | dan2 | symlink[laptop]: info is the wrong way to go |
| 02:56.44 | dan2 | symlink[laptop]: and we no longer support it |
| 02:57.03 | symlink[laptop] | I had other reasons for saying it, but nevermind |
| 02:57.03 | dan2 | symlink[laptop]: rfc2833 is the alternative to the rest |
| 02:57.30 | symlink[laptop] | I should switch back to my regular nick |
| 02:57.37 | *** join/#asterisk jaiger (~jaiger@65.86.55.130) |
| 02:58.15 | *** join/#asterisk dohja (robert@acs-24-154-249-207.zoominternet.net) |
| 02:58.18 | pbd | Snewpy- you still there? |
| 02:58.25 | pbd | I've got files for you, but can't dcc. |
| 02:58.33 | pbd | (Aren't firewalls just lovely?) |
| 02:58.35 | dohja | quick question about asterisk if anyone has a sec... |
| 02:58.40 | file[laptop] | dohja: just ask it |
| 02:58.53 | dohja | coo, i'm doin a project for school, and using asterisk as a solution for a pbx |
| 02:58.54 | pbd | Dohja- it works, mostly. |
| 02:59.18 | dohja | need to know if i'm implementing it to use SIP, would i still need SIP Proxy servers, Registrar servers etc... |
| 02:59.24 | dohja | or does asterisk take care of that |
| 02:59.25 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 02:59.31 | file[laptop] | asterisk does that |
| 02:59.41 | dohja | so one box will take care of all of taht |
| 02:59.43 | file[laptop] | you register to asterisk, you use asterisk to send calls, you use asterisk to receive calls... does it all |
| 02:59.56 | dohja | sweet, that just made things a lot easier :> |
| 03:00.41 | JerJer | ahh i sorta miss the days of standardized testing. I just came up with this crazy explaination of DUNDi |
| 03:00.42 | JerJer | BGP is to IP as DUNDi is to VoIP |
| 03:00.53 | denon | a pita? |
| 03:00.55 | denon | <G> |
| 03:00.58 | pbd | Dohja- what school is this that is enlightened enough to do a project using VoIP? |
| 03:01.11 | dohja | Pittsburgh Technical Institute |
| 03:01.25 | file[laptop] | Whois dohja |
| 03:01.27 | file[laptop] | bah |
| 03:01.28 | dohja | we didn't have to use VoIP, but we had to set up a phone system, with unlimited budget, for 3 companies |
| 03:01.32 | pbd | JerJer: The answer would be C. |
| 03:01.37 | JerJer | Indiana University (i think) Brian Capouch is a prof at some indiana collage teaching VoIP |
| 03:02.08 | pbd | Dohja: so I'm guessing this isn't a lab. |
| 03:02.10 | pbd | Rats. :) |
| 03:02.18 | dohja | heh, yeah sorry :P |
| 03:02.32 | pbd | JerJer- Wouldn't surprise me if it was IU- they're pretty far along. |
| 03:02.32 | dohja | they aren't stupid enough to let us go on irc at school |
| 03:02.44 | file[laptop] | I can get on IRC, it just takes SSHing through 3 boxes |
| 03:03.51 | Darwin35 | ok fixed |
| 03:04.02 | file[laptop] | Darwin35: version skew is a dingo |
| 03:04.51 | JerJer | a dingo ate your baby |
| 03:05.21 | pbd | Rhetorical question- if you're designing a mesh network, is it better to base the topology on a) a directed graph that's self learning and self healing, or b) whatever the stupid humans decide should be the best routing. |
| 03:05.44 | pbd | You're now on b. |
| 03:05.53 | Darwin35 | ok |
| 03:06.11 | libpcp | hi guys |
| 03:07.04 | JerJer | moo |
| 03:07.11 | libpcp | i this possible, from my ip phone i dialed exten 1 for musiconhold and when someone outside dial number extension is still can handle the call? |
| 03:07.46 | JerJer | hmm engrish? or chinglish? |
| 03:08.45 | libpcp | JerJer: what do you think is it possible? |
| 03:18.21 | tessier | Anyone know how I can adjust the amount of sidetone on a zap channel? |
| 03:18.32 | tessier | A cisco 7960 dialing out on these lines has way too much sidetone. |
| 03:19.12 | tessier | There is a page on sidetone on the wiki but nothing about how to change it |
| 03:19.55 | tessier | ARGH |
| 03:20.01 | tessier | My zap card just died again. |
| 03:20.03 | tessier | Piece of crap. |
| 03:20.16 | Moc | :( |
| 03:20.17 | paulc | libpcp: If you have call waiting turned on, sure. |
| 03:20.36 | tessier | This is my 4th day in the digium tech support hold queue. |
| 03:20.37 | *** join/#asterisk IJ-- (ianj@c-67-181-150-149.client.comcast.net) |
| 03:20.42 | tessier | WTF is going on over there? Anyone know? |
| 03:20.49 | tessier | A couple weeks ago I was able to just call up and get a tech. |
| 03:21.01 | paulc | I've heard they're pretty busy.. junky's been dealing with them on a PRI issue |
| 03:21.23 | tessier | But how could they have gone from answering calls right away to taking 4 days? |
| 03:21.26 | Moc | I heard of junky having a hardtime getting his problem looked at |
| 03:21.59 | Moc | tessier, My guess is they got a few request to out install system out of town, but dont have ennuf people in the back after |
| 03:21.59 | JerJer | junk-y may have found a real problem, Matt F is on it |
| 03:22.04 | IJ-- | Does anyone have any experience with the Quicknet Internet PhoneJACK card? I'm having a rather odd situation which I'm not sure if it's a configuration issue or a bug. |
| 03:22.18 | Moc | JerJer, strange you dont get the same problem |
| 03:22.18 | JerJer | quicknet is junk |
| 03:22.26 | JerJer | nope |
| 03:22.29 | Moc | I mean your have more pri than him |
| 03:22.31 | IJ-- | Well, it's all I have. |
| 03:22.39 | JerJer | ebay it |
| 03:22.51 | JerJer | let some other poor sole deal with it |
| 03:22.54 | tessier | Inbound calls always seem to work fine. Outbound calls work for a few minutes after a reboot and then the call goes dead and nobody can make any more outbound calls. |
| 03:23.00 | tessier | But inbound continue to work just fine. |
| 03:23.03 | tessier | Anyone heard of such an issue? |
| 03:23.16 | tessier | I was just on a dialed-out call and was talking and the line went dead. |
| 03:23.20 | Moc | tessier, try to rmmod zaptel and reload it again see if work again |
| 03:23.34 | tessier | Unfortunately the first time this happened the phone was in production and of course the CIO was talking to the CEO when it happened. :( |
| 03:23.48 | IJ-- | Well, what's your suggestion to use other than this? |
| 03:24.19 | Moc | yea that suck.. |
| 03:24.40 | *** join/#asterisk greg_work (~greg@ik-dynamic-66-102-67-60.kingston.net) |
| 03:25.13 | junk-u | ~seen kpflemming |
| 03:25.14 | jbot | i haven't seen 'kpflemming', junk-u |
| 03:25.20 | mishehu | anybody know what typical cost for a PRI is in chicago area? |
| 03:25.20 | junk-u | ~seen kpfleming |
| 03:25.21 | jbot | kpfleming <~chatzilla@wsip-68-14-253-125.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 1d 4h 8m 45s ago, saying: 'symlink[laptop]: the problem is that if the channels both have groups assigned (via SetGroup) the resulting channel ends up with both group assignments, even if that means you have two in the same category... that is bad'. |
| 03:25.47 | tessier | hmm |
| 03:26.01 | tessier | reloading the module brought it back...but as I talk the amount of sidetone seems to be decreasing |
| 03:26.33 | Moc | I have a problem that after a few week, my tdm400 card die |
| 03:26.35 | tessier | Wow, and I can't hear anything from the other end hardly either... |
| 03:26.44 | Moc | tessier, on FXS ? |
| 03:26.57 | tessier | Moc: FXO. This is 4 pots lines from the CO |
| 03:27.00 | Moc | I know FXS sound very bad and echo when my issue happen.. |
| 03:27.10 | Moc | FXO dont work anymore, but donno about active calls |
| 03:27.26 | Moc | tessier, the best guess we got so far, is that it the TDM400 is very sensitive to power.. |
| 03:27.45 | Moc | also you sure your TDM400 is fixed for using 1 FXO in port module 1 ? |
| 03:27.45 | tessier | Moc: If you are using it as FXO it does not need the extra power attached right? |
| 03:28.00 | Moc | tessier, not supose to ... but... |
| 03:28.07 | jaiger | anyone have iaxping.exe handy to test something for me? |
| 03:28.09 | file[laptop] | etter safe then sorry |
| 03:28.10 | file[laptop] | er better |
| 03:28.13 | tessier | My TDM400 has 4 FXO modules on it and they are all configured. |
| 03:28.37 | Moc | tessier, the TDM400 is a old board or it came with the 4 FXO on it ? |
| 03:28.47 | tessier | Came with the 4 fxo...new board |
| 03:28.56 | Moc | urm ok |
| 03:30.48 | Moc | tessier, it a TDM400 card issue then... I hope digium will fix it ... Im supose to send mine for replacement because of this problem |
| 03:30.55 | Moc | but more and more I see, it a general problel |
| 03:31.33 | *** join/#asterisk [1]funkknob (funkknob@203.223.34.133) |
| 03:31.39 | *** join/#asterisk putnopvut (~putnopvut@user-24-214-77-95.knology.net) |
| 03:31.55 | Moc | NEXT... |
| 03:34.44 | *** part/#asterisk IJ-- (ianj@c-67-181-150-149.client.comcast.net) |
| 03:35.25 | *** join/#asterisk _Eagle_ (nick@wasteland.net) |
| 03:36.31 | *** join/#asterisk _chad (~Chad@c-24-6-142-55.client.comcast.net) |
| 03:36.53 | _chad | hmm.. anyone had any experiance upgrading a 7940g to sip? |
| 03:37.15 | jaiger | I thought all the Gs were sip? |
| 03:37.35 | _chad | Hmm, that might explain some of the trouble i'm having |
| 03:38.09 | Moc | _chad, you need to say hi first... |
| 03:38.10 | _chad | would a 7940g actually have the 'g' on the front of the phone? |
| 03:38.11 | _chad | ha |
| 03:38.14 | _chad | Mac, Hi :) |
| 03:38.43 | Moc | _chad, 7960g have icon buttons, doesnt have text on the buttons |
| 03:38.49 | jaiger | _chad: dunno, I only have polycom and grandstream |
| 03:38.56 | denon | _chad: if the "Servics" etc buttons have pictures on them .. its a G |
| 03:39.03 | denon | services/messages/settings |
| 03:39.07 | _chad | yep, has icons |
| 03:39.13 | denon | then its probably a G |
| 03:39.17 | denon | check the back if you're concerned |
| 03:39.26 | Moc | G normally mean Global |
| 03:39.28 | _chad | the settings in the menu said it was a g.. i just bought it as a 7940 |
| 03:39.40 | Moc | all the new phone are G |
| 03:39.40 | _chad | so do i need to do a sip conversion on these then? |
| 03:39.51 | *** join/#asterisk z_smurf (~z_smurf@jamlikheten-77-70.ip-pluggen.com) |
| 03:39.53 | *** part/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 03:39.53 | z_smurf | Hallo |
| 03:39.54 | _chad | Mac, well these have seen a bit of use i'm guessing |
| 03:39.57 | Moc | probably, you see SIP in the configuration menu ? |
| 03:40.06 | z_smurf | Where did MYSQL_FRIENDS go in channels/Makefile?? |
| 03:40.26 | paulc | Anyone fancy joining us for a game of acro? The most fun you can have with a keyboard? |
| 03:40.30 | denon | G also supports normal POE I thought |
| 03:40.37 | _chad | Well.. I was monkeying with them, got the tftp setup w/ the proper build in there configured as best as I can tell.. now it seems to log in and errors out looking for a p0s30202.sbn file |
| 03:40.39 | denon | heh .. more acro spam |
| 03:40.41 | _chad | then resets.. kinda seems looped |
| 03:40.58 | paulc | denon.. c'mon.. you know you wanna.. |
| 03:41.01 | Moc | denon, that what they say, but cant find the datasheet |
| 03:41.16 | denon | /kick #asterisk paulc Here's some fun I can have with a keyboard! |
| 03:41.35 | paulc | LOL.. ok, ok.. |
| 03:42.05 | D1ng0 | sup guys |
| 03:42.18 | z_smurf | anyonw know where MYSQL_FRIENDS went? It was included in channels/chan_sip.c a few months ago |
| 03:42.19 | Moc | hi |
| 03:42.32 | jaiger | anyone have iaxping.exe handy? |
| 03:42.54 | D1ng0 | Im about to take a sledge hammer to this * box >;\ |
| 03:42.56 | _chad | Where exactly would it say SIP in the config menu? |
| 03:43.42 | _chad | i reset the factory settings on here.. hrm |
| 03:43.52 | _chad | *taps phone with a pen* |
| 03:44.19 | file[laptop] | z_smurf: it's been replaced with realtime in CVS head, but it is still present in CVS stable |
| 03:44.49 | _chad | bootload id is pc0303010001, thats non sip yah? |
| 03:45.05 | z_smurf | file[laptop] "realtime" .. Any docs how to use that? |
| 03:45.18 | file[laptop] | z_smurf: look on voip-info |
| 03:45.26 | z_smurf | ok |
| 03:45.54 | file[laptop] | all everyone needs to know is on there |
| 03:45.56 | D1ng0 | ya'll realize the wiki info for IAX2 trunking is wrong ??? its backwards.... |
| 03:45.57 | file[laptop] | and more |
| 03:46.29 | D1ng0 | either that or D1ng0 is dislexic |
| 03:47.35 | _chad | okay heres a simple question, it says in the firmware versions menu, Version: 5.0(3.0)... which bin should i be upgrading this to? |
| 03:47.40 | z_smurf | file[laptop] I only find documentation about MYSQL_FRIENDS there too... .Where is "realtime"?... |
| 03:48.24 | file[laptop] | z_smurf: it's there, believe me |
| 03:50.01 | D1ng0 | where ahem, i believe "client side" should say "server side" and "server side" should say "client side" in the iax.conf examples... |
| 03:50.45 | z_smurf | file[laptop] res_config? |
| 03:50.50 | D1ng0 | though hence the reversal of "host=" in both fixes the issue |
| 03:51.10 | file[laptop] | z_smurf: you're gonna make me give you a direct URL aren't you? |
| 03:51.27 | file[laptop] | http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20Sip |
| 03:51.38 | file[laptop] | then at the bottom, "See Also" |
| 03:51.46 | z_smurf | :) |
| 03:51.55 | D1ng0 | here bkw_ bkw_ bkw_ ..... ive got a cookie for you...! :) |
| 03:52.31 | mishehu | anybody know a good PRI provider in the chicago area? |
| 03:53.35 | D1ng0 | boy it got awful quiet in here |
| 03:54.07 | mishehu | there was a hacker who had a dog, and D1ng0 was his name-oh... D...1...n...g...0... |
| 03:54.16 | mishehu | sorry, couldn't help but sing |
| 03:54.17 | D1ng0 | ROFL |
| 03:54.34 | mishehu | especially since nobody wants to give me a professional opinion. |
| 03:54.45 | D1ng0 | mishehu, whassup ? need sumthin |
| 03:54.57 | D1ng0 | ahh a PRI provider |
| 03:55.28 | pbd | Only PRI's I use around here (Mount Prospect) is from SBC and Sprint. |
| 03:55.32 | pbd | Haven't had trouble with either one. |
| 03:55.41 | pbd | But I woudln't say they're cheap. |
| 03:58.43 | *** join/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
| 03:59.21 | *** part/#asterisk putnopvut (~putnopvut@user-24-214-77-95.knology.net) |
| 04:01.59 | D1ng0 | Current Array Status : create - 94% complete YAY!!! 6 more hours im sure |
| 04:02.36 | *** join/#asterisk luckyali (~lukyali@203.81.196.167) |
| 04:02.55 | luckyali | morning ppl |
| 04:05.56 | luckyali | ~adn |
| 04:05.57 | jbot | extra, extra, read all about it, daily asterisk news is http://www.sineapps.com/news.php for html and http://www.sineapps.com/rssfeed.php for rss |
| 04:06.02 | *** join/#asterisk Umaro (~umaro@c-24-22-76-14.client.comcast.net) |
| 04:06.56 | *** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca) |
| 04:08.46 | sudhir492 | Anyone using TDM400 for PSTN interface? |
| 04:09.07 | Umaro | Hey guys, getting a really weird problem on my E1 where I get SAMBE requests and what not from the network side, and I see * sending back acknowledgements, etc, but the other side is never recieving them. |
| 04:09.18 | mishehu | pbd: how much are the PRI's running you each? |
| 04:09.33 | mishehu | sudhir492: I am. |
| 04:09.41 | mishehu | tdm04b, 4 fxo ports |
| 04:09.55 | Umaro | I've been working on this problem for about a month, and i'd rather not anymore.. if anyone can solve this for Me, I will pay you $500 USD via paypal. |
| 04:10.18 | sudhir492 | mishehu: Do they work well or you are having problem, e.g. echo/call drop anything? |
| 04:10.25 | mishehu | Umaro: can't be much of help, as we don't use e1's around here. |
| 04:10.44 | Moc | same here :( |
| 04:10.48 | cypromis | e1 PRI ? |
| 04:11.06 | sudhir492 | mishehu: I am planning to use TDM04B too (4 ports FXO) |
| 04:11.11 | mishehu | sudhir492: you have echo issues with any analog equipment. I've gotten them adjusted pretty well though. |
| 04:11.22 | mishehu | sudhir492: I'd hold up on purchasing if you can though... |
| 04:11.31 | mishehu | if you haven't already bought one |
| 04:11.41 | mishehu | there is one annoying bug they seem to have... |
| 04:12.10 | mishehu | after about 2 to 6 weeks of usage, they'll just suddenly stop working. you'll have to stop asterisk, unload the kernel mods, reload them, and then restart asterisk. |
| 04:12.36 | z_smurf | file[laptop] Fantastic.. Its working ;) |
| 04:12.39 | z_smurf | Thanx a lot |
| 04:12.41 | mishehu | they are beta testing a fix right now though, and I am going to be getting some of the replacement modules. |
| 04:13.08 | file[laptop] | z_smurf: yw |
| 04:13.22 | sudhir492 | mishehu: I placed an order today, but they have not yet shipped. Can still cancel them. But I need to prepare my system by Tuesday. Do I have other alternative? |
| 04:13.43 | z_smurf | file[laptop] Even the cdr is working as before.. Wow.. :-) ... |
| 04:13.43 | mishehu | sudhir492: contact them and ask them for a card with the beta modules... |
| 04:14.08 | mishehu | they might have a few. |
| 04:14.35 | mishehu | sudhir492: even if they don't, the card will work. it just might need a cron script to handle all that stuff on off-peak hours. |
| 04:14.43 | mishehu | once every week or 2. |
| 04:14.44 | sudhir492 | mishehu: Hmm. Putting beta version in production - it is like lesser evil |
| 04:15.12 | mishehu | your other option is to get a channel bank and a t100p... |
| 04:15.46 | sudhir492 | mishehu: Can I get by by rebooting the machine itself every weekend ? |
| 04:16.39 | luckyali | anyone knows about Metaboo - Asterisk Game ? |
| 04:17.15 | sudhir492 | mishehu: How much can I expect to pay for a channel bank? |
| 04:17.52 | mishehu | sudhir492: like I said, you don't need to restart it completely. just write a script that stops asterisk, modprobe -r 's the wcfxo/wctdm modules, wait a couple seconds, reload the modules, rerun ztcfg, then restart asterisk. |
| 04:17.58 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 04:18.15 | mishehu | I'm not sure how much and which kind of channel bank to use. I know other ppl have used them though. |
| 04:18.41 | mishehu | luckyali: first games for your cellphone, now games for your pbx? |
| 04:19.09 | iris___ | are there any good sources for add on agi or cisco services addons? |
| 04:20.44 | sudhir492 | mishehu: I sure can write a script to reset the card, say every night. Will that be sufficient for proper functioning ? |
| 04:21.07 | luckyali | Metaboo is an Asterisk game developed by Twisted and drumkilla, yet to be included into * |
| 04:21.41 | *** join/#asterisk john8675309tmmm (~chatzilla@207.177.124.87) |
| 04:22.19 | *** join/#asterisk Moc___ (~mochouina@modemcable212.49-80-70.mc.videotron.ca) |
| 04:23.04 | john8675309tmmm | is there any easy way to connect asterisk into my exsiting analog panasonic pbx? |
| 04:23.35 | Moc___ | john8675309tmmm, we do it with Sipura SPA-2000 |
| 04:23.40 | Moc___ | 2 port FXS port |
| 04:23.40 | mishehu | sudhir492: every night is unnecessary, but you can do it. I'd do it once every 2 weeks. |
| 04:23.51 | Moc___ | but you can use a TDM400 card too |
| 04:24.57 | jaiger | john8675309tmmm: I do it with some FXO interfaces |
| 04:25.00 | sudhir492 | mishehu: ok. I have noticed similar problem with a t100p card too. One of the asterisk boxes locks up randomly after couple of weeks. |
| 04:25.19 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 04:25.36 | |Vulture| | Moc_: do you have the newest SIP image for the IP500s? |
| 04:25.42 | |Vulture| | I heard they just released a new one |
| 04:25.42 | jaiger | sudhir492: the whole box or just the card? |
| 04:25.43 | datareactor | i will zaphfc will work with Fedora http://www.voip-info.org/wiki-Asterisk+zaphfc+install |
| 04:26.07 | sudhir492 | mishehu: The t100p card is old. On the other hand, I have t405 (quad T1 card) on another machine which runs flawlessly |
| 04:26.08 | john8675309tmmm | I could also do that with grandstream devices or do they need to be fxo ports? |
| 04:26.14 | jaiger | and is it use related? |
| 04:27.58 | jaiger | |Vulture|: what's the new firmware fix/add? |
| 04:28.22 | *** join/#asterisk _Vile (~vile@90.b160.bendtel.net) |
| 04:28.30 | |Vulture| | jaiger: not a clue |
| 04:28.58 | sudhir492 | Actually, the box with t100p card gets around 1000 calls a day, and after couple of week or so, I notice that I cannot send any calls through that. I am not sure what exactly is happening, and I dont have time to debug this as this invariably happens on a Monday morning, when I have no time to debug, I just ask the guy to restart the machine |
| 04:29.34 | mishehu | sudhir492: is that machine up-to-date? |
| 04:29.42 | mishehu | as far as software and drivers are concerned |
| 04:30.55 | sudhir492 | mishehu: up-to-date? Definitely not. The machine has asterisk version CVS-HEAD-07/27/04 |
| 04:33.00 | *** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca) |
| 04:33.10 | sudhir492 | mishehu: But it runs great otherwise, hence I have not bothered to do anything with it other then rebooting the box every other weekend :-) |
| 04:34.41 | mishehu | personally, I'd not think of a machine that needs a reboot every 2 weeks as "running great" |
| 04:34.54 | mishehu | I normally set up machines and forget about them... |
| 04:36.46 | *** join/#asterisk brent21 (~brent21m@24.152.235.191.res-cmts.snh.ptd.net) |
| 04:38.10 | sudhir492 | mishehu: I agree with you. My own machine which I use for providing virtual PBX service for businesses, has been running for 185 days, and asterisk since Oct 21. But this particular machine is at a client site |
| 04:40.12 | bkw_ | must not take many calls |
| 04:40.41 | sudhir492 | mishehu: I meant the machine where asterisk locks up once in a while. When it is running, it meets their funtional requirements well |
| 04:40.53 | sudhir492 | bkw_: are you talking to me? |
| 04:40.59 | bkw_ | no the wall.. |
| 04:41.02 | D1ng0 | uhhh whos got the new ip500 rom and sip images ? are they available ? |
| 04:41.07 | bkw_ | who do you think i'm talkin to |
| 04:41.15 | *** join/#asterisk GarryH (~guangyaoc@S0106000f66a57e29.ed.shawcable.net) |
| 04:41.58 | BoRiS | bkwhore :) |
| 04:42.06 | bkw_ | hey horis |
| 04:42.11 | *** join/#asterisk wolfson (hehe@cpe-24-196-251-110.man.nc.charter.com) |
| 04:42.17 | D1ng0 | here bkw_ bkw_ bkw_ ..... ive got a cookie for you...! :) |
| 04:42.21 | bkw_ | BoRiS how do you downgrade glibc? |
| 04:42.29 | bkw_ | :P |
| 04:42.36 | D1ng0 | bkw_, by installing a BSD |
| 04:42.39 | BoRiS | ack |
| 04:42.50 | bkw_ | D1ng0 no dear |
| 04:43.01 | bkw_ | but its stupid lame |
| 04:43.05 | Moc | ordered my firefox NYT Ad poster and my Fox ;) |
| 04:43.06 | sudhir492 | D1ng0: I am using ip500, around 40 of them. my image is about a month or so old. But works great. |
| 04:43.47 | Moc | D1ng0, what version do you have ? |
| 04:44.13 | D1ng0 | sudhir492, ip version 1.3.1 and rom 2.5.0 |
| 04:44.21 | Moc | that will work just fine.. |
| 04:45.22 | Moc | sudhir492, they are great phone arnt they ? |
| 04:45.31 | Moc | I gota make a web GUI to configure them thought.. |
| 04:45.37 | D1ng0 | well yeah it is working.... no fixes in the 1.3.4 ??? |
| 04:45.56 | Moc | D1ng0 check polycom website to see what being fixed and if it affect you |
| 04:46.12 | Moc | D1ng0, you have issues with your phone ? |
| 04:47.12 | sudhir492 | bkw_: A simple yes/no would have been sufficient. You didn't have to get offended. My stable asterisk box gets around 2500 calls on weekdays, 60% of them ANSWERED, some to voicemail, and some not answered |
| 04:48.07 | Moc | sudhir492, bkw_ being in bad mood since last year, (first time ive meet him) |
| 04:48.34 | Brixius | anyone here using cdr_mysql? |
| 04:48.34 | Moc | sudhir492, dont take it personal |
| 04:48.34 | sudhir492 | D1ng0: yest bootrom 2.5.0 |
| 04:48.37 | Moc | sudhir492, UNLESS you didnt say hi when you join.. then your screwed .. |
| 04:48.42 | Brixius | that might be able to answer a question or 2? |
| 04:48.59 | sudhir492 | Moc: :-) |
| 04:49.23 | D1ng0 | Moc, nope seems im good, nothing important that affects me in 1.3.4 and my boot rom is current |
| 04:49.23 | Moc | Brixius, I was, but moved to postgresql |
| 04:49.23 | Brixius | quit |
| 04:49.28 | Brixius | opps, wrong window |
| 04:49.29 | Moc | then your in buisness ;) |
| 04:49.39 | sudhir492 | Mos: Yes, polycom ip500 are great phones! Cheaper than SNOM 190 too. |
| 04:49.42 | *** join/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
| 04:49.49 | jcollie | hello all |
| 04:49.58 | D1ng0 | Moc, heh ill be in buisiness when i get this damn IAX2 trunk working right :P |
| 04:50.00 | Moc | sudhir492, does snom have as much feature as polycom ? |
| 04:50.06 | Brixius | I hate when I do that, luckly I wasn't typing a password. :o |
| 04:51.05 | Moc | damn .com NS server get updated damn fast now, why use a DynDNS !! ;) |
| 04:51.58 | D1ng0 | ok im going to sleep, ill fight with this IAX2 trunk tomorrow |
| 04:52.04 | jcollie | is there such a thing as a cordless phone that would talk 5.8Ghz or 2.4Ghz (_not_ 802.11b) to base station and base station would talk SIP to *? |
| 04:52.29 | sudhir492 | Moc: Yes, featurewise they are quite comparable. But I do not like the look and feel of SNOM 190 at all. The handset keeps falling of the cradle. Polycom's speakerphone/microphone is a lot better too. On top of that, I could not connect plantronics headset without an amplifier to SNOM. |
| 04:52.59 | Moc | hha... so good thing I never tryed snom then hehe |
| 04:53.09 | D1ng0 | night guys |
| 04:53.13 | sudhir492 | Moc: Whereas plantronics headsets work great with Polycom without any amplifier. |
| 04:53.17 | Moc | i love beable to configure dialplan per extensions |
| 04:53.32 | Moc | and having different voicemail extention |
| 04:54.00 | sudhir492 | jcollie: Yes, use any ATA with any standard cordless phone :-) |
| 04:54.04 | D1ng0 | Moc, heh go fix my IAX2 trunks then :) |
| 04:54.20 | Moc | and... I just saw today, you can set in the directory on the phone, that if call is from UserA, I get different ringtone, or I can ignore or transfer the call or ... ;) too cool |
| 04:54.22 | jcollie | sudhir492: bah, i knew someone was going to suggest that |
| 04:54.40 | Moc | D1ng0, I dont use trunk. got ennuf bandwidth rightnow |
| 04:54.49 | *** join/#asterisk ccfiel (~ccfiel@210.213.141.18) |
| 04:54.55 | ccfiel | hello pll.. |
| 04:55.02 | Moc | hi |
| 04:55.06 | D1ng0 | Moc, not a bandwidht issue, more a remote location connection |
| 04:55.06 | ccfiel | do i need to register all my extension? |
| 04:55.12 | Moc | if you wish |
| 04:55.20 | ccfiel | in my iax.conf? |
| 04:55.23 | Moc | no need to use the trunk feature in iax |
| 04:55.28 | sudhir492 | jcollie: you sure show signs of brilliance sometimes :-) |
| 04:55.30 | Moc | you can do what ever you want ccfiel ;) |
| 04:55.57 | D1ng0 | Moc, uhhh if i wanted server A to connect to server B for remote office dial out what would i use then ? |
| 04:56.00 | ccfiel | Moc: i am confuse of how asterisk handle conf... |
| 04:56.26 | sudhir492 | D1ng0: Use IAX trunking. Works well for me |
| 04:56.40 | Moc | D1ng0, you can use IAX, but you dont need to set the channel as trunk |
| 04:56.50 | Moc | trunk will only save bandwidth |
| 04:57.44 | sudhir492 | Moc: How much do you pay for Polycom ip500? |
| 04:58.08 | pbd | G'night all. Thanks, snewpy. |
| 04:58.08 | blitzrage | ccfiel: basically each channel has a configuration file (iax.conf, sip.conf, h323.conf, mgcp.conf, voicemail.conf, etc....) and these are all tied together in the extensiosn.conf (the dialplan) |
| 04:58.14 | D1ng0 | okay well ive got servA connected/registered with ServB but when a user from servA dial 1XXXNXXNXXX i keepm getting unathorized user request on ServB |
| 04:59.54 | D1ng0 | Moc, and from ServA i get Call rejected by 209.114.214.3: No authority found |
| 05:00.42 | D1ng0 | and heres the exten => _1NXXNXXXXXX,1,Dial(IAX2/remote:hidden@209.114.214.3/${EXTEN}@local) |
| 05:00.45 | Moonwick | so. since my asterisk server is suffering from hard drive problems, I've started thinking about asterisk failover... is there such a thing? |
| 05:00.46 | ccfiel | blitzrage: yes.. i have define in iax.conf my firefly account.. with a context X and i define an internal account with a context y but when i call my firefly number it will go to context y... |
| 05:00.49 | blitzrage | where's http://bugs.digium.com/karma_hallofshame.php |
| 05:00.50 | blitzrage | :) |
| 05:00.59 | Moonwick | my (granted, light) googling didn't turn up much |
| 05:01.05 | bjohnson | sudhir492: $200 |
| 05:02.09 | D1ng0 | Moonwick, yes its in the Wiki |
| 05:02.25 | sudhir492 | bjohnson: I get at $175 each at present. I am looking for a cheaper source. |
| 05:03.04 | D1ng0 | Moonwick, start here http://www.voip-info.org/tiki-index.php?page=Asterisk%20High%20Availability%20Solutions |
| 05:03.18 | bjohnson | I can't wait untill bluetooth is standard in voip phones |
| 05:03.28 | D1ng0 | anyway... im off to sleep |
| 05:03.30 | D1ng0 | night |
| 05:03.44 | bjohnson | sudhir492: in quantity or retail? where? |
| 05:05.17 | sudhir492 | I usually purchase in quantities of 10. But can purchase in quantites of 5 too, at the same price without paying shipping cost. |
| 05:05.17 | |Vulture| | Moc: you around? |
| 05:05.43 | |Vulture| | anyone ever have an IP-500 freeze with the MWI lit, the polycom background on, doesn't boot just sits at that |
| 05:05.55 | Brixius | this is kindof off topic, but anyone good at postgresql it's spitting out "user root does not exist" no matter what I try, (I'm giving up on mysql) |
| 05:06.40 | bjohnson | Brixius: you'd be better off in #postgresql |
| 05:06.48 | Brixius | thanks |
| 05:06.53 | Brixius | I'll try that. |
| 05:06.58 | sudhir492 | bjohnson: From tritech, phone number 18008913388 |
| 05:07.25 | *** join/#asterisk AvengerMoJo (~alex@61.149.1.237) |
| 05:07.45 | AvengerMoJo | hi |
| 05:08.37 | AvengerMoJo | where can I find the doc for asterisk ( about how to use those features? ) |
| 05:09.01 | *** join/#asterisk phsdshft (phsdshft@dialup-4.156.81.197.Dial1.Boston1.Level3.net) |
| 05:09.26 | |Vulture| | oh nice... IP-500 recovered itself :P |
| 05:09.46 | bjohnson | sudhir492: thnx. found their website from that. |
| 05:10.30 | Moc | |Vulture| Im here |
| 05:10.41 | Moc | never saw that |
| 05:11.40 | |Vulture| | yea its fixed now |
| 05:11.51 | |Vulture| | its one of those still in h323 mode |
| 05:12.24 | *** join/#asterisk Alric (~nbowyer@masq.hyperusa.com) |
| 05:12.37 | Moc | oh ok |
| 05:13.05 | |Vulture| | Moc: this time on my new order I made sure to get SIP rofl |
| 05:13.14 | |Vulture| | I orderd 40 of them... no way Im doing this to 40 phones |
| 05:13.21 | Moc | lol Good ;) |
| 05:13.25 | Moc | ip 500 or 600 ? |
| 05:13.28 | |Vulture| | 500 |
| 05:13.34 | |Vulture| | I still need to get a 600 |
| 05:13.43 | Moc | yea h323 is in 500 only |
| 05:14.05 | Moc | 600 is nice, if you need the feature |
| 05:14.18 | |Vulture| | well I wouldn't mind the 600s for admin work |
| 05:14.56 | |Vulture| | I use a 7960 right now |
| 05:15.15 | |Vulture| | good phone.. but I do like the 500s better |
| 05:15.47 | Moc | |Vulture|, yea, I just discovered the magic of the phone directory of the polycom ... |
| 05:16.05 | |Vulture| | I haven't used it yet |
| 05:16.13 | |Vulture| | you got a -phone.cfg example I can check out? |
| 05:16.27 | Moc | |Vulture|, for the directory ? |
| 05:16.29 | |Vulture| | I had the whole xml center on the 7960s |
| 05:16.30 | |Vulture| | yea |
| 05:16.47 | Moc | well directory .cfg file will be created on your FTP once you add them on the phone |
| 05:17.07 | |Vulture| | oh, you add them from the phone.. I thought you meant like a master directory |
| 05:17.27 | |Vulture| | upgrading from h323 is so brutal! |
| 05:17.29 | Moc | no, on the phone, the feature you have like differemt ringtone |
| 05:17.42 | badc{}l{}r | WARNING[15622]: pbx.c:1995 ast_pbx_run: Timeout, but no rule 't' in context 'sip'. What does it mean? |
| 05:17.50 | Moc | and beable to ignore, or transfer automaticly a call |
| 05:18.12 | Moc | Im having a BIG PHP/Apache problem.. |
| 05:18.21 | file[laptop] | badc{}l{}r: Timeout, but no rule 't' in context 'sip'. |
| 05:18.30 | Moc | I moved my site from a RH9 to a RHEL3 ... and now all my accents show as square .. |
| 05:18.58 | *** join/#asterisk digitize1 (~joehamoud@ip24-251-149-184.ph.ph.cox.net) |
| 05:19.01 | |Vulture| | Moc: Im running FC3 on this new server |
| 05:19.10 | |Vulture| | and I cant get mysql working right |
| 05:19.23 | Moc | |Vulture|, Fedora core always caused me problem.. |
| 05:19.37 | Moc | I use TaoLinux (RHEL3 recompiled version) |
| 05:19.41 | Moc | work very well |
| 05:20.50 | Moc | hhhaaa fixed .. good |
| 05:20.54 | |Vulture| | kinda late now Im pretty much done configuring |
| 05:21.06 | |Vulture| | everything works but my mysql |
| 05:21.14 | Moc | connect to a remote mysql DB |
| 05:21.18 | Moc | for the moment |
| 05:21.42 | ccfiel | hello.. |
| 05:22.34 | ccfiel | can somebody explain..to me..im confuse of how asterisk sequence of action are taken in context..:( |
| 05:23.20 | |Vulture| | Moc: I only use it for CDR comes in handy to track telemarketers |
| 05:23.27 | |Vulture| | offfice managers LOVE CDR |
| 05:23.49 | Moc | oh yea ;) |
| 05:23.51 | |Vulture| | "You mean I can see how many mins, and how many calls my employees make?" ... SOLD |
| 05:24.02 | Moc | hehe |
| 05:24.36 | |Vulture| | nortar is vanashing and in place a nice dell server :) |
| 05:25.21 | Moc | |Vulture|, I just got myself a Dell PowerEdge 1850, not bad machine |
| 05:26.13 | *** join/#asterisk Urban_Viking (~xx@CA03F8FA.adsl.mana.pf) |
| 05:26.42 | Moc | a nice dual Xeon 2.8, 1gig ram, 2x36gig hd.. |
| 05:26.44 | *** join/#asterisk shaZwaz (~lukyali@203.81.196.167) |
| 05:27.01 | Moc | alittle over kill for a 5 user * PBX but hehe, dell has a 25% off deal ;) |
| 05:27.11 | Urban_Viking | Since I upgrade from 1.0.0 to 1.0.3 , my Zyxel P2002 box wont work any more , any idea ? |
| 05:27.40 | |Vulture| | nice |
| 05:27.43 | |Vulture| | yea |
| 05:27.51 | |Vulture| | I use SC420slol |
| 05:27.57 | |Vulture| | for small offices they work wonders |
| 05:28.17 | |Vulture| | Urban_Viking: you upgraded zaptel right? |
| 05:28.26 | |Vulture| | oh sorry I read that wrong |
| 05:29.00 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 05:29.01 | Moc | DDR2 ram cost too much !!! |
| 05:29.03 | Urban_Viking | I upgrade asterisk |
| 05:30.06 | Moc | ok sleep now.. cya |
| 05:30.08 | |Vulture| | very true |
| 05:30.13 | |Vulture| | but 512 seems to be fine |
| 05:30.25 | |Vulture| | its under $500 for the server |
| 05:30.26 | |Vulture| | night |
| 05:31.40 | bjohnson | there still really isn't a practical, cost effective way to use a cell phone as a roaming extension is there? |
| 05:33.16 | |Vulture| | unless your on nextel and have free incoming :) |
| 05:34.20 | shaZwaz | bjohnson: wait until WiMax arrives |
| 05:37.17 | *** join/#asterisk jradford (~jradford@hoss.npl.com) |
| 05:37.26 | jradford | evening all |
| 05:38.21 | |Vulture| | sup jr |
| 05:38.32 | shaZwaz | hi jradford |
| 05:38.54 | jradford | just finished eating my dinner out of a colostomy bag, how about you? |
| 05:40.07 | |Vulture| | nope can't top that |
| 05:40.16 | |Vulture| | sounds like you had an MRE |
| 05:42.05 | jradford | I listen to way too much talk radio I think |
| 05:43.21 | bjohnson | shaZwaz: wimax? |
| 05:43.52 | |Vulture| | http://www.wimaxforum.org/ |
| 05:44.27 | BoRiS | ugh.... 802.11n |
| 05:45.41 | shaZwaz | bjohnson: http://www.wimaxxed.com |
| 05:46.24 | badc{}l{}r | sorry guy, i'm newbie. I always getting this msg, how should i fix it? WARNING[15622]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call |
| 05:47.13 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.we.client2.attbi.com) |
| 05:47.59 | Wi_Fi | anyone have access to a fax machine |
| 05:48.13 | jradford | I do wi_fi |
| 05:48.20 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 05:48.31 | Wi_Fi | can you send me a test fax |
| 05:48.39 | Urban_Viking | Since I upgrade from 1.0.0 to 1.0.3 , my Zyxel P2002 box wont work any more , any idea ? |
| 05:48.40 | jradford | sure, whats your fax # |
| 05:48.57 | shaZwaz | I had no luck with outgoing fax |
| 05:49.55 | shaZwaz | txfax doesn't transmit data |
| 05:49.55 | Wi_Fi | window jradford |
| 05:51.02 | bjohnson | I don't get it. I can see how wimax may replace cdma or gsm .. but how would it allow a cell phone to act as a voip handset? |
| 05:51.32 | Brixius | badc: I think I got that when I was using a budgetone phone and my codec's wern't configured correctly, but that was a long time ago, and I don't remember the spicifics. |
| 05:52.14 | shaZwaz | bjohnson: who said you ill need a cell phone with Wimax |
| 05:52.18 | |Vulture| | anyone know if cdr_addon_mysql doesn't work with 4.1? |
| 05:52.45 | Wi_Fi | fuqin awsome |
| 05:52.48 | Wi_Fi | WOW |
| 05:52.52 | Wi_Fi | im blown away |
| 05:53.00 | Wi_Fi | thnx jradford |
| 05:53.03 | Wi_Fi | it worked |
| 05:53.18 | Wi_Fi | you sent a fax and it went straight to voicemail |
| 05:53.38 | jradford | what are you using for handling inbound faxes? |
| 05:53.47 | Wi_Fi | * |
| 05:53.56 | jradford | what fax handler? opendsp? |
| 05:54.08 | Wi_Fi | * spand |
| 05:54.17 | jradford | cool |
| 05:54.29 | jradford | one day when I get bored I'll have to try and set that up |
| 05:54.31 | shaZwaz | rxfax works fine |
| 05:54.43 | jradford | rxfax runs python correct? |
| 05:54.50 | shaZwaz | wonder anyone tried txfax ? |
| 05:54.52 | jradford | err: written in python |
| 05:54.56 | Wi_Fi | now .. how can i send a fax without a fax machine? |
| 05:55.18 | shaZwaz | Wi_Fi: there is a same app called txfax |
| 05:55.31 | shaZwaz | works with the .call files |
| 05:55.33 | Wi_Fi | yes .. its runnin |
| 05:55.39 | *** join/#asterisk rvhi (~rv@mail.o-matrix.org) |
| 05:55.41 | rvhi | hi |
| 05:55.54 | Wi_Fi | is there a windows client |
| 05:56.06 | rvhi | i have some polycom 500 and isdn pri card |
| 05:56.18 | rvhi | i keep getting echo on polycom phones |
| 05:56.25 | Wi_Fi | Im runnin a successful AMP installation |
| 05:56.31 | rvhi | tried everything suggested on wiki |
| 05:56.36 | rvhi | still no luck |
| 05:56.42 | rvhi | anyone have suggestions? |
| 05:56.54 | shaZwaz | like what rvhi |
| 05:57.03 | *** join/#asterisk mikes2277 (~mike@wireless-206.222.58.98.bwsys.net) |
| 05:57.23 | rvhi | on the polycom phone, when i talk, i can hear myself |
| 05:57.39 | rvhi | the latency from polycom to asterisk is only about 10ms |
| 05:57.43 | shaZwaz | I mean what did you try to fix it ? |
| 05:57.46 | Moonwick | on the telepathyphone, I can hear myself think. |
| 05:58.03 | mikes2277 | ok, dumb question... how does one check their voice mail messages without using VoiceMailMain i.e. pressing * and password while listening to your own greeting |
| 05:58.14 | shaZwaz | Moonwick: thats cool |
| 05:58.31 | rvhi | adjust zapata.conf |
| 05:58.33 | *** join/#asterisk ptblank (~MURDER1@68-233-237-22.anhmca.adelphia.net) |
| 05:58.54 | rvhi | echocancel=256 |
| 05:59.00 | rvhi | echotraining=yes |
| 05:59.02 | shaZwaz | try echotraining=128 |
| 05:59.15 | shaZwaz | echocancel=32 |
| 05:59.39 | mikes2277 | anyone know how? |
| 05:59.57 | rvhi | then restart asterisk, or there is more involved? |
| 06:00.12 | mikes2277 | nope, thats it |
| 06:00.17 | shaZwaz | mikes2277: it can be done in the dialplan |
| 06:00.31 | mikes2277 | how, I keep trying but nothing is working for me |
| 06:00.51 | shaZwaz | show application VoiceMailMain on CLI |
| 06:01.01 | mikes2277 | what I want is the customer to be able to break out of their own message in VoiceMail(mailbox) just like a cell phone... |
| 06:01.01 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:01.30 | mikes2277 | I know, but that requier that the VM box be known ahead of time |
| 06:01.55 | mikes2277 | and that the customer call a specific ext for that |
| 06:02.35 | rvhi | same |
| 06:02.42 | rvhi | same echo |
| 06:02.54 | rvhi | what else should i try? |
| 06:02.54 | mikes2277 | increae to 64 or 128 |
| 06:03.01 | mikes2277 | it will go away |
| 06:03.18 | mikes2277 | asterisk software echo can ist not very good |
| 06:03.31 | shaZwaz | rvhi: did you try the uncommenting the Aggressive_Suppressor in zaptel |
| 06:04.06 | shaZwaz | mikes2277: what are you trying to do ? |
| 06:04.10 | mikes2277 | http://www.voip-info.org/wiki-Asterisk+VoiceMail infers that you can dial out from voicemail |
| 06:04.25 | rvhi | pretty much the saem |
| 06:04.34 | rvhi | yes, i did uncommenting |
| 06:05.07 | rvhi | after make;make install; i did insmod zaptel;insmod ;ztcfg |
| 06:05.17 | mikes2277 | http://bugs.digium.com/bug_view_page.php?bug_id=0000156 this one says you can login to mailbox during greeting....but not how |
| 06:05.35 | Urban_Viking | Since I upgrade from 1.0.0 to 1.0.3 , my Zyxel P2002 box wont work any more , any idea ? |
| 06:06.06 | rvhi | i tried 128 and even 256, nothing changed |
| 06:06.29 | mikes2277 | it wont work past 128, are you running the latest libpri and zaptel from cvs? |
| 06:06.45 | rvhi | no, it is 1.0.0 |
| 06:06.50 | mikes2277 | try cvs |
| 06:07.08 | rvhi | it is production system, a little hesitate to change it now |
| 06:07.16 | mikes2277 | otherwise you can do what we did, give up cause its too flakey and get a Lucent TNT |
| 06:07.43 | mikes2277 | We had a lot of echo issues where it would just not work on a random channel for no apparent reason |
| 06:08.23 | rvhi | can we adjust txgain or rxgain? |
| 06:08.38 | *** join/#asterisk [1]funkknob (funkknob@203.223.34.133) |
| 06:08.42 | shaZwaz | low values help sometimes |
| 06:09.13 | rvhi | low on both? |
| 06:09.15 | shaZwaz | rvhi: you calling from zap-sip ? |
| 06:09.27 | rvhi | from pstn to sip |
| 06:09.30 | mikes2277 | that is just for volume |
| 06:09.33 | rvhi | same result the other directy |
| 06:09.45 | TheLostOn | anyone alive with a running dundi |
| 06:09.48 | mikes2277 | where is your PRI from? |
| 06:09.57 | rvhi | i tried to change txgain and rxgain, no much change |
| 06:10.00 | rvhi | time warner |
| 06:10.01 | mikes2277 | phone company? |
| 06:10.18 | rvhi | go through adtran 550 |
| 06:10.19 | mikes2277 | they can turn on echo can on their side too if you ask REAL nicely |
| 06:10.33 | mikes2277 | the adtran *should* have echo can capability |
| 06:10.45 | Wi_Fi | jradford where do you live? |
| 06:11.05 | mikes2277 | which side is the echo heard on? The PSTN or VoIP side? |
| 06:11.17 | shaZwaz | Wi_Fi: let me know if you get txfax working |
| 06:11.21 | rvhi | sip side |
| 06:11.34 | mikes2277 | That is the PSTN's sides problem then, not asterisk |
| 06:11.50 | Wi_Fi | shaZwaz i dont have a fax machine to test |
| 06:11.57 | mikes2277 | Echo issues are from the opposite side that you can hear it on |
| 06:12.06 | shaZwaz | send it to jradford |
| 06:12.20 | mikes2277 | Seems like Asterisk is fine, check your adtran or further down the pipe |
| 06:12.20 | rvhi | if i call from t-mobile cell, no echo |
| 06:12.33 | rvhi | but verizon landline |
| 06:12.38 | rvhi | sprint cell, |
| 06:12.41 | mikes2277 | Yup, PSTN problem |
| 06:12.48 | Brixius | yeah!!!!, postgres logging is working, I can connect with my windows box to query the db, I'm happy. |
| 06:12.58 | rvhi | is it adtran or tw? |
| 06:13.11 | mikes2277 | could be either, who owns the Adtran? |
| 06:13.18 | rvhi | we |
| 06:13.19 | Wi_Fi | shaZwaz how do i send a fax |
| 06:13.32 | mikes2277 | login to it and see if you can find echo can settings for it |
| 06:13.57 | rvhi | on it now... searching... |
| 06:14.01 | mikes2277 | it seems like some carriers arnt doing it for you so you need to do it for yourself OR get TW to do it for you. |
| 06:14.24 | shaZwaz | Wi_Fi: use a .call file |
| 06:14.37 | mikes2277 | its generally best if the echo can happens as close as possible to each endpoint but sometimes you have to do what you have to do.... |
| 06:14.52 | mikes2277 | and echo can in the middle |
| 06:14.56 | Wi_Fi | shaZwaz example? |
| 06:14.59 | mikes2277 | its not as good but it helps |
| 06:15.11 | mikes2277 | wifi, use G711 |
| 06:15.27 | mikes2277 | it wont be reliable but for a few pages |
| 06:15.34 | rvhi | also got tw support on the line |
| 06:15.34 | shaZwaz | Wi_Fi: http://www.voip-info.org/tiki-index.php?page=Asterisk%20fax |
| 06:15.46 | rvhi | what question should i ask? |
| 06:15.46 | mikes2277 | then network qos almost always ruins it |
| 06:15.57 | mikes2277 | can they turn on echo can on your pri |
| 06:16.19 | mikes2277 | I assume its a pri going into your adtran which is pealing off channels to you, right? |
| 06:16.34 | mikes2277 | the other channels are used for data? |
| 06:16.53 | rvhi | all voice |
| 06:16.57 | rvhi | some number goes live |
| 06:16.57 | shaZwaz | rvhi: which adtran |
| 06:17.03 | rvhi | some number to dev |
| 06:17.04 | mikes2277 | PRI though right? |
| 06:17.09 | rvhi | adtran 550 |
| 06:17.13 | rvhi | yes |
| 06:17.24 | rvhi | so no echo, i guess |
| 06:17.25 | shaZwaz | is there any echo can on that one ? |
| 06:17.29 | rvhi | it is transparent |
| 06:17.33 | shaZwaz | I dont think so |
| 06:17.45 | rvhi | no echo on pstn side |
| 06:17.51 | shaZwaz | its in 850 I guess |
| 06:17.51 | TheLostOn | anyone alive with a running dundi setup |
| 06:18.03 | rvhi | 850 can do echo? |
| 06:18.12 | mikes2277 | ask TW for G.168 echo can to be turned on |
| 06:18.22 | shaZwaz | I heard it has a builtin echocanceler |
| 06:18.45 | shaZwaz | CAC Adit too is good |
| 06:18.51 | mikes2277 | if they have a DMS or 5E it will do it |
| 06:19.03 | mikes2277 | CAC is cool but $$$$$ |
| 06:19.21 | *** join/#asterisk CoderCR (~creyna@ip68-8-11-127.sd.sd.cox.net) |
| 06:19.25 | CoderCR | hello all |
| 06:19.31 | shaZwaz | rvhi: check with ztmonitor |
| 06:19.43 | mikes2277 | Echo can really needs to happen on verizon's side though, the echo can on TW will only mildly help |
| 06:19.58 | mikes2277 | so anyone know how to check voicemail from within the greeting? |
| 06:20.03 | mikes2277 | * codes dont work |
| 06:20.25 | rvhi | i tried ztmonitor, but don't know what i can do |
| 06:20.29 | CoderCR | does anyone here have a 480i? |
| 06:20.33 | shaZwaz | mikes2277: the patch doesn't work ? |
| 06:20.52 | mikes2277 | the patch is supposed to be in cvs over 6 months ago |
| 06:21.17 | shaZwaz | play with rxgain, txgain , echocancel, echotraingin, echocanwhenbridged values |
| 06:21.30 | shaZwaz | and check on ztmonitor |
| 06:21.59 | rvhi | it is moving bar, how to read it? |
| 06:22.10 | mikes2277 | looking in the source I see where * is supposed to let it out but it isnt working |
| 06:23.03 | mikes2277 | 0 will get you out to the operator ext of "o" but then I don't know which mailbox they were just in.... |
| 06:23.12 | mikes2277 | unless there is a var or something.... |
| 06:23.18 | shaZwaz | rvhi: each * represents a value |
| 06:23.39 | rvhi | i changed txgain |
| 06:23.59 | rvhi | i can't see any change on the moving bar |
| 06:24.02 | shaZwaz | if there is echo , speaking on tx it will appear on rx too |
| 06:24.14 | CoderCR | Does anyone here have a Sayson 480i? |
| 06:24.25 | mikes2277 | didnt think they were shipping yet |
| 06:26.39 | rvhi | yes |
| 06:26.50 | rvhi | tx is about 15-20 |
| 06:26.56 | rvhi | tx is about 3-5 |
| 06:27.39 | rvhi | so if i change txgain to -10, i should get better results? |
| 06:27.50 | rvhi | also rxgain to -10? |
| 06:27.53 | shaZwaz | mikes2277: I still didn't get it , log into mailbox during greeting ? |
| 06:28.59 | TheLostOn | anyone alive with a running dundi setup |
| 06:29.14 | CoderCR | mikes2277: they are for developers and beta testers |
| 06:29.15 | shaZwaz | rvhi: does ur set do any silence suppression ? |
| 06:29.15 | rvhi | thx guy, appreciate all your help |
| 06:29.23 | rvhi | no |
| 06:29.44 | shaZwaz | try with a soft client |
| 06:29.52 | *** join/#asterisk Shido6 (~shido@d57-87-253.home.cgocable.net) |
| 06:30.27 | rvhi | what will that do? |
| 06:30.40 | rvhi | i thought * doesn't allow VAD |
| 06:30.48 | shaZwaz | to check if the set has something to do with it |
| 06:30.50 | rvhi | it breaks the rtp? |
| 06:31.47 | *** part/#asterisk CoderCR (~creyna@ip68-8-11-127.sd.sd.cox.net) |
| 06:32.24 | rvhi | sorry, i have to run now. thanks for all your help |
| 06:32.29 | rvhi | really appreciate it |
| 06:32.35 | Shido6 | :) |
| 06:32.45 | Shido6 | whats up? |
| 06:34.12 | *** join/#asterisk jtodd (~jtodd@66.165.175.229) |
| 06:34.41 | *** join/#asterisk _chad (~Chad@c-24-6-142-55.client.comcast.net) |
| 06:36.41 | _chad | hi, anyone w/ any experiance upgrading 7940g's to work w/ asterisk? |
| 06:37.00 | |Vulture| | you need the sip images |
| 06:39.54 | _chad | yeah i snagged them down |
| 06:40.20 | _chad | have 7.3, but i'm having a hard time understanding exactly what needs to happen here... i've tried several attempts with no luck |
| 06:40.58 | _chad | i have the phone directly connected to a nic on this machine, its running tftp, w/ a stash of files in the directory.. it seems to log into tftp but the phone just dosnt take the new config |
| 06:40.59 | *** part/#asterisk AvengerMoJo (~alex@61.149.1.237) |
| 06:41.37 | _chad | continues to try and open the callmanager ip (or at least thats what the phone says) |
| 06:42.45 | _chad | incredibly frustrating |
| 06:43.15 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 06:43.17 | brc_ | ~seen twisted |
| 06:43.18 | jbot | twisted is currently on #asterisk. Has said a total of 47 messages. Is idling for 4h 21m 26s |
| 06:44.07 | Atacomm | anyone have thoughts on the Zhone channel banks? |
| 06:45.03 | Brixius | anyone know how I can tell what libraries a module is tring to use? |
| 06:45.23 | znoG | ldd? |
| 06:45.46 | Brixius | thanks |
| 06:46.18 | Atacomm | or anyone use an Carriar Access Access Bank I? |
| 06:47.15 | Shido6 | yes |
| 06:47.38 | _chad | mm. |
| 06:48.01 | _chad | does the dhcp feature on the 7940 typically not work? |
| 06:48.05 | _chad | or am i just smoking rocks |
| 06:48.23 | Atacomm | shido: yes to who and regarding what? lol |
| 06:50.01 | badc{}l{}r | Where can i get more details information on configuring call forwarding between sip client |
| 06:51.29 | znoG | www.voip-info.org |
| 06:52.06 | *** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc) |
| 06:52.24 | JerJer | smokin rocks! |
| 06:52.27 | badc{}l{}r | it too brief |
| 06:56.39 | shaZwaz | Atacomm: are the zhone cb still in production ? |
| 06:57.01 | _chad | alright if my phone is looking for a .sbn file does that mean its running a current rev? |
| 06:57.04 | Atacomm | doubt it, cant find it on their site. i'm looking on ebay |
| 06:57.14 | _chad | even before that, how can u tell if a 79xx phone is in sip mode? |
| 06:57.19 | shaZwaz | I use an Adit 600 |
| 06:57.29 | shaZwaz | works like a charm |
| 06:57.52 | Shido6 | if you provision your iaxy with a bad ip .... like the ip of your router |
| 06:58.09 | Atacomm | i'm looking for some cheap gear, just for lab testing |
| 06:58.12 | Shido6 | or public ip that exists.... |
| 06:58.21 | Shido6 | how do you reprovision the iaxy ? |
| 06:58.25 | _chad | shindo6, was that directed for me? |
| 06:58.30 | Shido6 | anyone :) |
| 06:58.32 | shaZwaz | ebay is a good place to look |
| 06:58.35 | _chad | lol |
| 06:59.50 | Wi_Fi | sheet cant get a call file to execute |
| 06:59.50 | Atacomm | yup, thats where i'm looking... looking at Zhone, CAC (AB1 and AB2s), ec |
| 06:59.52 | Atacomm | *etc |
| 07:00.40 | *** join/#asterisk pchenry (~pchenry@firewall1.enetonline.net) |
| 07:03.36 | _chad | holy mackarel the "updating software" message! |
| 07:03.42 | _chad | blessed be the lord |
| 07:03.47 | Atacomm | LOL |
| 07:03.56 | Atacomm | well |
| 07:04.05 | Atacomm | you arent out of the woods until it actually works :) |
| 07:04.22 | _chad | i'm just happy to see it do something different |
| 07:04.27 | _chad | really takes my breath away lol |
| 07:04.33 | pchenry | 7960? |
| 07:04.44 | _chad | 7940g |
| 07:04.49 | pchenry | ahh |
| 07:04.55 | Atacomm | lol, yeah, i remember i spent several nights once fighting with those phones.... some versions of the firmware you cant upgrde straight to the latest |
| 07:05.10 | _chad | yeah thats what i've been banging on |
| 07:05.11 | Atacomm | due to a TFTP overflow glitch in the cisco firmwares |
| 07:05.13 | pchenry | what code did U start wth ? |
| 07:05.15 | _chad | i was trying to goto 7.3 |
| 07:05.17 | _chad | i think 5 |
| 07:05.20 | pchenry | yeah |
| 07:05.22 | _chad | went to 6.3 okay |
| 07:05.23 | pchenry | should work |
| 07:05.43 | Atacomm | pchenry: any experience with channel banks? |
| 07:05.55 | pchenry | yeah ... what kind ? |
| 07:06.01 | pchenry | CAC ? |
| 07:06.40 | Atacomm | looking at a Carrier Access AB1 FXS.... there are a group of them for $65 on ebay, just for lab testing.... we need various devices to test our upcoming t1 card against.... |
| 07:06.49 | _chad | goin for 7.3 now |
| 07:07.01 | *** join/#asterisk Umaro (~umaro@c-24-22-76-14.client.comcast.net) |
| 07:07.05 | pchenry | yeah .. CAC is good |
| 07:07.30 | Atacomm | ok, voip-info seemed to just have ? marks next to comments on them |
| 07:07.58 | pchenry | what are you gonna try and do ? |
| 07:08.00 | Atacomm | they seemed to not like the AB-1, but of course the price is great for lab gear.... i just dont want to buy something that ends up not being a good device to test against |
| 07:08.17 | _chad | damn |
| 07:08.22 | _chad | ProtocalApplicationInvalid |
| 07:08.33 | pchenry | hmmm |
| 07:08.44 | _chad | something w/ the cnf? |
| 07:08.48 | Atacomm | trying to put together a few devices to test against our upcoming T1/E1 card.... my goal is to basically verify that fxs or fxo signalling in the drivers is written properly |
| 07:08.56 | pchenry | did you put the right filename in the DEFAULT file ? |
| 07:09.01 | pchenry | same as in OS |
| 07:09.08 | *** join/#asterisk kimosabe (~kimo@201.133.195.215) |
| 07:09.12 | _chad | pchenry, yeah that seemed to take fine |
| 07:09.27 | pchenry | are you watching the TFTP server ? |
| 07:09.28 | kimosabe | does any one know a voice over ip provider with 956 area code |
| 07:09.43 | kimosabe | laredo tx provided area code |
| 07:09.57 | Umaro | Hey guys, how do I make * provide a dialtone? |
| 07:10.09 | pchenry | ata ... the CAC boxes will sqeeze tight to the standard .... so they will be good or that |
| 07:10.19 | _chad | yeah |
| 07:10.20 | pchenry | provide dialtone to what ? |
| 07:10.24 | Atacomm | ok great |
| 07:10.37 | kimosabe | which service provider is the best vonage ? |
| 07:10.39 | pchenry | or = for ata |
| 07:10.40 | pchenry | oops |
| 07:10.57 | pchenry | whats it looking for chad |
| 07:11.14 | pchenry | be some cant finds in there |
| 07:11.16 | _chad | actually my log viewer is crashing everytime i try and view the log.. not a good night |
| 07:12.26 | _chad | hm. |
| 07:12.37 | _chad | cant seem to find the logfile :( |
| 07:12.53 | Umaro | pchenry: a zap channel |
| 07:13.01 | pchenry | whats the OS file have in it |
| 07:13.15 | _chad | P003-07-3-00 |
| 07:13.26 | pchenry | and now whats in the DEFAULT |
| 07:13.33 | pchenry | for the image name |
| 07:13.53 | _chad | ha your the man :) |
| 07:13.59 | pchenry | wait |
| 07:14.02 | pchenry | what was it |
| 07:14.20 | _chad | was pos3-04-4-00 |
| 07:14.23 | pchenry | ahhh |
| 07:14.28 | pchenry | thatll do it |
| 07:14.38 | _chad | awesome lets give that a tweak |
| 07:14.44 | pchenry | U need POS3-07-3-00 |
| 07:14.51 | pchenry | NOT P00 |
| 07:15.13 | pchenry | an S and not a 0 |
| 07:15.28 | _chad | hmm |
| 07:15.31 | _chad | POS or P0S ? |
| 07:15.37 | _chad | P0S3-07-3-00? |
| 07:16.01 | *** join/#asterisk rumba (~ropawa@cs68201148-205.sw.rr.com) |
| 07:16.21 | pchenry | zero |
| 07:16.24 | pchenry | 0 |
| 07:16.26 | _chad | k |
| 07:16.33 | *** join/#asterisk Yoda-BZH (~yoda-bzh@pro.game.admin.at.yoda-bzh.net) |
| 07:16.37 | *** join/#asterisk implicit (~implicit@ip68-7-149-247.sd.sd.cox.net) |
| 07:16.37 | pchenry | sorry |
| 07:16.55 | Yoda-BZH | le bonjour je vous souhaite |
| 07:16.56 | _chad | no prob, i really appreciate the help |
| 07:18.02 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:18.52 | Umaro | Hey guys, how do I make * provide a dialtone to a zap channel? |
| 07:19.20 | Umaro | Do I have to do fxs signalling? |
| 07:19.37 | implicit | yeah |
| 07:20.03 | _chad | hmm |
| 07:20.11 | _chad | pchenry still bailing on me.. any other ideas? |
| 07:20.33 | pchenry | need to see those tftp logs |
| 07:20.34 | Umaro | implicit: no other way? :( |
| 07:20.48 | implicit | Umaro: what do you mean? |
| 07:20.55 | pchenry | nothing in the SIPmac.cnf file is ther |
| 07:20.58 | pchenry | about the image name |
| 07:21.06 | _chad | nope |
| 07:21.06 | Brixius | anyone know if there's a better tapi driver then asttapi, one that doesn't cause outlook to give error's on every other launch? |
| 07:21.11 | Umaro | implicit: well, i'm providing a E1 to this siemens phone system |
| 07:21.13 | _chad | and i did track down the tftp log |
| 07:21.22 | kimosabe | does www.voxilla.com/iconnectehere lock up youre spa if you use the configuration wizard ? |
| 07:21.32 | implicit | sup Atacomm |
| 07:21.32 | pchenry | what files you have in there ? |
| 07:21.33 | Umaro | and I need it to hear a dialtone |
| 07:21.37 | pchenry | all of the files from cisco ? |
| 07:21.50 | implicit | what kind of siemens switch do you have? |
| 07:22.00 | _chad | yeah i yanked down a couple different dists i found.. should it work with JUST the cisco files? |
| 07:22.04 | implicit | this is a class4 switch you are talking about? |
| 07:22.06 | Atacomm | implicit: not much, just bidded on a CAC AB1 FXS... i hope it includes everything, LOL.... looking to use it as a test unit for our cards |
| 07:22.10 | Umaro | no clue really, why? :) |
| 07:22.12 | _chad | the latest dist i'm trying has tons of pcm's and raws and so on |
| 07:22.23 | pchenry | should have the 7-03 files .... |
| 07:22.23 | implicit | Atacomm: i dont know anything about tose fxs' |
| 07:22.27 | pchenry | an OS file |
| 07:22.34 | pchenry | and a SIPDEFAULT |
| 07:22.40 | Urban_Viking | does anyone know about any provider giving french number to collect calls ? |
| 07:22.41 | pchenry | dont need anything else |
| 07:22.41 | Atacomm | neither do i, its an older model, so no reference on the CAC website |
| 07:23.57 | pchenry | want me to setup my tftp and you can point to it ? |
| 07:24.52 | _chad | pchenry that would rock :D |
| 07:24.54 | *** join/#asterisk Umaro (~umaro@c-24-22-76-14.client.comcast.net) |
| 07:25.02 | pchenry | lemme loook .. |
| 07:25.04 | _chad | oop i'm doing the striped down version now |
| 07:25.07 | _chad | looks like it might have worked |
| 07:25.14 | _chad | just those files you mentioned |
| 07:26.28 | libpcp | anyone is using sipura 3000 ? |
| 07:26.52 | _chad | sweet, major progress |
| 07:27.15 | _chad | pchenry, dialed out just fine but got the dreaded dead after the person picked up the line (the 7940 went dead) |
| 07:27.39 | _chad | asterisk reported a max retry timeout also |
| 07:27.59 | libpcp | how do i change the ip address of the sipura3000 using the telephone handsey |
| 07:28.03 | shaZwaz | Wi_Fi: whats up ? |
| 07:28.06 | _chad | pchenry, if you have your tftp i'd love to grab it just so I can have something to compare to config wise |
| 07:28.09 | libpcp | s/handsey/handset |
| 07:29.01 | *** join/#asterisk Trimor (~www@202.179.135.215) |
| 07:29.35 | maksim | hello folks |
| 07:29.47 | maksim | is this right? -> http://pastebin.ca/3098 |
| 07:31.21 | Trimor | hi can ne body help me out regarding Extensions.conf and Dial cmd |
| 07:31.36 | maksim | I don't think I did it correct because X.Y.Z.B can still make a call... :\ |
| 07:33.00 | shaZwaz | maksim: you missed the second lable in Gotoif |
| 07:33.52 | shaZwaz | GotoIf(Condition?label1:label2) |
| 07:34.20 | maksim | shaZwaz, I wish to XYZB can't call in this exten... what I must correct? |
| 07:34.42 | shaZwaz | correct ur gotoif first |
| 07:36.54 | maksim | exten => _XXXX.,1,GotoIf($[${OH323_RADDR} = X.Y.Z.B]?3:2) |
| 07:37.13 | shaZwaz | k |
| 07:37.15 | maksim | X.Y.Z.B don't call now :) |
| 07:37.48 | maksim | shaZwaz, thanks |
| 07:37.56 | shaZwaz | ;) |
| 07:38.21 | shaZwaz | Wi_Fi: where are you ? |
| 07:38.29 | rvhi | hi |
| 07:38.42 | rvhi | i am back with more about my echo issue |
| 07:38.44 | shaZwaz | het rvhi |
| 07:38.49 | rvhi | it is definitely * |
| 07:39.00 | rvhi | i have a cisco router with pri card |
| 07:39.06 | shaZwaz | did you try it with a soft client |
| 07:39.07 | rvhi | it connects to adtran |
| 07:39.14 | rvhi | don't have any echo issue |
| 07:39.31 | rvhi | haven't tried soft client yet. |
| 07:39.40 | rvhi | but since the only difference is cisco and * |
| 07:39.52 | rvhi | cisco absolutely no echo |
| 07:40.06 | shaZwaz | give it a try |
| 07:40.18 | shaZwaz | may be ur polycom is the culprit |
| 07:40.19 | rvhi | ok |
| 07:40.30 | rvhi | well, it is the same polycom |
| 07:40.38 | rvhi | one line to *, one line to cisco |
| 07:41.37 | shaZwaz | rvhi: what rx/tx gain are you using |
| 07:41.47 | rvhi | default 0, 0 |
| 07:41.56 | rvhi | tried rx=-4.5 |
| 07:42.03 | rvhi | sorry |
| 07:42.05 | shaZwaz | adtran has echo issues |
| 07:42.12 | rvhi | tx=-4.5, rx=0 |
| 07:42.28 | rvhi | with cisco, it also has adtran |
| 07:42.42 | rvhi | also it is just passthrough |
| 07:42.49 | rvhi | not like 850 |
| 07:43.40 | shaZwaz | ok start with the echocancelwhenbridged=yes ? |
| 07:43.59 | rvhi | y |
| 07:44.47 | shaZwaz | what machine ? |
| 07:46.26 | rvhi | it is a dell 1750 |
| 07:46.50 | rvhi | echo only becomes an issue if latency is high |
| 07:46.59 | shaZwaz | did you uncoment the MMX directive in ztconfig.h |
| 07:47.00 | rvhi | from phone to *, only 10ms |
| 07:47.20 | rvhi | so could it be pci bus slow it down |
| 07:47.27 | rvhi | not yet |
| 07:47.29 | rvhi | let me check |
| 07:49.37 | *** join/#asterisk kongnamool (~sexton@67.113.12.162) |
| 07:51.13 | kimosabe | does any one have a number i can call 2 test my service provider thanks |
| 07:53.01 | Brixius | call domino's see if they answer. |
| 07:53.47 | Trimor | i have this configuration running |
| 07:53.49 | Trimor | [line2user] |
| 07:53.49 | Trimor | exten => _X., 1, Dial(vpb/1-2/${VPBPAUSE}${EXTEN},8) |
| 07:53.49 | Trimor | exten => _X., 2, Hangup |
| 07:54.24 | Trimor | this line is in our private exchange i want to shift my call to some other service provider |
| 07:54.36 | kimosabe | brixius do u have a dominos number i can call im in mexico sorry |
| 07:54.40 | Trimor | for that i have to pass first of all 9 to call outside of my office |
| 07:54.49 | kimosabe | area code and all |
| 07:54.54 | Trimor | so im trying to use 9, my number but will |
| 07:55.01 | rvhi | there is no ztconfig.h |
| 07:55.06 | Trimor | but it starts giving me error |
| 07:55.19 | Trimor | i mean congestion ring |
| 07:55.37 | rvhi | zconfig.h? |
| 07:55.43 | Brixius | I think there's a domino's in cancun, I ordered from them once when I was there. |
| 07:55.58 | Brixius | haha |
| 07:55.58 | Trimor | ne ideas ? |
| 07:56.21 | Brixius | Actually I wish I was there now. it;s cold here in Minnesota |
| 07:56.36 | *** join/#asterisk Dega (~justin@196.22.202.2) |
| 07:57.17 | shaZwaz | rvhi: err its zaptel.h |
| 07:58.12 | JerJer | 27* in Michigan |
| 07:58.56 | shaZwaz | rvhi: its zconfig.h |
| 08:00.13 | shaZwaz | #define CONFIG_ZAPTEL_MMX |
| 08:00.27 | Dega | hmm |
| 08:00.38 | *** join/#asterisk Hanuza (~HanuzGran@193.226.188.67) |
| 08:00.45 | Dega | should i get capi cards for an interface to the pstn ? |
