| 21:31.33 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 21:31.33 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 21:31.40 | vader-- | like two weeks ago you said you were going to be doing work on the protocol |
| 21:31.50 | Qwell[] | vader--: yeah, when I get time I do |
| 21:32.14 | woolbeo | I've got a problem with voicemail. Some user's will record their unavailable, busy, and greeting, but * still plays the default voicemail message. I checked /var/spool/asterisk/voicemail/context/mailbox#/ and the files are there, but * isn't recognizing them. |
| 21:32.36 | Qwell[] | woolbeo: You have to specify them, in the Voicemail() call... |
| 21:32.44 | Qwell[] | ie Voicemail(1234|u) |
| 21:32.57 | woolbeo | Qwell, already done that |
| 21:33.27 | LostFrog | Is there any way of telling whether ztdummy is working correctly without using meetme? |
| 21:33.50 | woolbeo | Qwell, what I mean is that it plays the default unavailable message, and it only hapens with some mailboxes. |
| 21:40.19 | enjay- | I have a handytone 286 that was registered for several days now all of a sudden it wont register its on the network I can reach it but status shows "Registered: No" |
| 21:41.00 | *** join/#asterisk BryanFE2 (n=chatzill@ip72-194-209-224.sb.sd.cox.net) |
| 21:42.18 | BryanFE2 | can someone please advise as to whether the Manager API can be used to place a call, wait for a tone (not a DTMF tone), and then continue (i.e. play an audio file)? I can't figure out the "wait for tone" step, not sure if the Manager API even supportsit. |
| 21:43.23 | enjay- | Is there an asterisk CLI command to force a registration or registration renew? |
| 21:43.54 | file | enjay-: you want to force a remote device to reregister? |
| 21:44.03 | eKo1 | enjay-: restart? |
| 21:44.16 | enjay- | I've rebooted it 100 times already notta.. |
| 21:44.49 | enjay- | I want to see if I can send it something to get it to re-register or something Im kind of at a loss here been troubleshooting it for a few hours.. |
| 21:45.01 | mrdigital | anyone here used the zoom 5801 ata |
| 21:45.11 | eKo1 | enjay-: power-cycle the device? |
| 21:45.19 | file | hehe you can't... it's up to the device to register, even if it were possible it would have to be registered with Asterisk to begin with in order to send a packet to it |
| 21:45.25 | enjay- | <enjay-> I've rebooted it 100 times already notta.. |
| 21:45.43 | eKo1 | Oh, I though you meant *. |
| 21:45.46 | enjay- | yea I kinda figured just wierd that its not even attempting to register.. |
| 21:45.46 | eKo1 | *thought |
| 21:45.47 | fgwaller | BryanFE2: You can do that but the easiest thing would be a combination of AMI and Dialplan... |
| 21:45.56 | *** join/#asterisk reticent (n=reticent@207-232-98-114.ip.van.radiant.net) |
| 21:45.59 | LostFrog | Exactly 100 times?? Have you tried 123 times? |
| 21:46.06 | eKo1 | lol |
| 21:46.09 | reticent | heh |
| 21:46.13 | BryanFE2 | fgwaller - I'll look at AMI, but in the meantime, what Manager API do I use? |
| 21:46.18 | file | or... 1234! |
| 21:46.39 | LostFrog | Remind me to change the combination on my luggage. |
| 21:46.42 | BryanFE2 | oh sorry you meant the Asterisk Manager Interface (I'm already there). |
| 21:46.48 | enjay- | heh |
| 21:46.50 | fgwaller | BryanFE2: Which ones are there? |
| 21:47.56 | fgwaller | BryanFE2: You have to use AMI. AGI is not a manager API... |
| 21:48.25 | BryanFE2 | I'm using AMI already (I presume you mean the manager interface) |
| 21:48.53 | fgwaller | did you read the the wiki on voip-info.org? |
| 21:51.10 | woolbeo | any ideas on why asterisk 1.2.9.1 will play vm-into instead of the recorded unavailable message on some mailboxes? |
| 21:55.05 | *** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243) |
| 21:57.10 | Zodiacal | can polycom's not park calls? |
| 21:57.14 | Zodiacal | with asterisk? |
| 21:58.12 | pdtwork | I have problems with it from time to time |
| 21:58.18 | reticent | does asterisk support the "a=direction:active" SIP/SDP parameter? |
| 21:58.20 | rowter | whats a best way to have musicandhold so it does not consumes lots of cpu, with native? |
| 21:58.24 | reticent | i can't find any information on the web regarding this |
| 21:58.26 | pdtwork | if you don't forward before the message is done playing sometimes it will hangup |
| 21:58.39 | *** join/#asterisk wangster (n=wangster@static-64-201-170-129.ptr.terago.ca) |
| 21:59.07 | BryanFE2 | does anyone know if there is an asterisk app which I could use from the dialplan, to wait for a specific tone from the other party? (not a DTMF tone) |
| 22:00.24 | *** part/#asterisk woolbeo (n=woolbeo@toby.stoneflytech.com) |
| 22:04.39 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 22:04.49 | Qwell[] | rowter: make it use the formats the call uses, so it doesn't have to transcode |
| 22:05.17 | rowter | Qwell, like for ulaw sox fpm-sunshine.wav -t ul -r 8000 -b -c 1 fpm-sunshine.pcm? |
| 22:05.31 | rowter | Qwell, like for ulaw sox fpm-sunshine.wav -t ul -r 8000 -b -c 1 fpm-sunshine.g711? |
| 22:06.54 | carl0s- | Hi guys |
| 22:07.10 | Qwell[] | rowter: .ul |
| 22:07.24 | carl0s- | What effect would adjusting the "RTP Packet Length" for g.711 have? (In my SIP gateway)? It's currently 20ms. Would reducing it to 10ms have any positive effect on delay? |
| 22:07.59 | Lyfe | is there a document online about codec qualities, etc, that puts it in plain, boring menial language for someone like me to understand? :) |
| 22:08.14 | rowter | Qwell, ok let me try out .ul |
| 22:09.22 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:09.37 | Zodiacal | any word if asterisk will implement shared appearances? |
| 22:09.51 | syzygyBSD | I am getting a ton of zombie processes from my agi script and I can't figure out why or how I can clean them up |
| 22:10.44 | syzygyBSD | anyone know how to clean up zombies? besides shooting their heads |
| 22:11.01 | LostFrog | reboot? lol |
| 22:11.25 | syzygyBSD | ya.. isn't a real fix though |
| 22:15.46 | LostFrog | I was just being an asshole. |
| 22:16.32 | wangster | when I do a show channels I have a channel in there thats been there at least a couple weeks. |
| 22:16.43 | Corydon-w | syzygyBSD: What language are you using to code your AGI? |
| 22:16.50 | *** join/#asterisk pirast (n=martin@p508B2B3A.dip0.t-ipconnect.de) |
| 22:16.50 | syzygyBSD | python |
| 22:16.56 | Corydon-w | Eww |
| 22:17.18 | pirast | hi guys, i want to have ( ) in my asterisk useragent.. how can i do that? |
| 22:17.28 | syzygyBSD | lol, I have coded a huge one in it before, had this problem then too, fixed it somehow but didn't write it down |
| 22:18.37 | *** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se) |
| 22:19.58 | pirast | syzygyBSD was ist a reply to my question? :-) |
| 22:20.20 | syzygyBSD | no, was a reply to Corydon's Eww |
| 22:21.02 | pirast | syzygyBSD: ah ok |
| 22:22.02 | LostFrog | pirast: smell sarcasm? |
| 22:22.32 | LostFrog | ohh.. nm |
| 22:23.04 | pirast | LostFrog hehe no :-) |
| 22:24.23 | Qwell[] | russellb: Well, that was easy... |
| 22:27.52 | LostFrog | I think I need a threaded IRC client in my old age.. I can't keep track of convos like I used to. |
| 22:31.56 | *** join/#asterisk mitcheloc (n=mitchelo@mail.coversant.net) |
| 22:32.56 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 22:32.56 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 22:39.58 | *** join/#asterisk adorah (n=Administ@87.68.169.141.cable.012.net.il) |
| 22:40.14 | *** part/#asterisk LostFrog (n=LostFrog@c-24-3-160-119.hsd1.pa.comcast.net) |
| 22:41.04 | *** join/#asterisk tbdba (i=tbdba@ip-216-97-188-180.valornet.com) |
| 22:41.27 | *** part/#asterisk tbdba (i=tbdba@ip-216-97-188-180.valornet.com) |
| 22:41.32 | adorah | Hi is ther anyone who has some knowledge about telephones and gain? |
| 22:41.41 | *** part/#asterisk jarrod (i=jarrod@juniperyour.net) |
| 22:41.46 | Qwell[] | adorah: people knowing about phones here? nah |
| 22:42.05 | adorah | well..this is law tech staff:) |
| 22:42.13 | adorah | low= |
| 22:42.19 | fgwaller | maybe then |
| 22:42.21 | Qwell[] | What's your question? |
| 22:42.46 | adorah | I connected two phones to the same extension |
| 22:43.02 | adorah | one regular phone works well |
| 22:43.19 | fgwaller | and the other one? |
| 22:43.27 | adorah | the other- a cordless one not very new sounds to loww.. |
| 22:43.47 | Qwell[] | a cordless phone should have a volume setting... |
| 22:43.54 | adorah | any way to increase volume of the cordless? |
| 22:44.04 | adorah | that is not a new one.. |
| 22:44.10 | Qwell[] | Get a new one :) |
| 22:44.24 | adorah | they r both connected to grandstream 496 ATA |
| 22:44.45 | adorah | do u think a neww dect will sound better? |
| 22:44.57 | Qwell[] | dect? |
| 22:45.13 | adorah | the digital 2.4 cordless ones.. |
| 22:45.19 | Qwell[] | oh...yeah, probably |
| 22:46.08 | adorah | coz if the user will spend 100$US for a new one and it won't get better..well..he wouldn't like it.. |
| 22:46.20 | fgwaller | thats in impedance issue, very low tech |
| 22:46.59 | adorah | yeah sure..but I guess there is nothing I can do to improve by config? |
| 22:47.20 | adorah | is ther any booster I can put on the cable? |
| 22:47.39 | fgwaller | when ever you have multiple phones on the same line, the one with the lowest impedance has the highest gain... |
| 22:48.23 | fgwaller | some phones have an adjustable gain, but most $5.50 will not |
| 22:48.36 | adorah | ok than how can I solve it? coz b4 it was connected directly to the CO same way and was ok.. |
| 22:49.32 | fgwaller | were they connected to regular CO line before? |
| 22:49.38 | adorah | yup.. |
| 22:50.06 | fgwaller | CO surely has a different impedance than that ATA then... |
| 22:50.14 | adorah | and they like it that way and not seperate extension.. |
| 22:50.35 | syzygyBSD | does asterisk send a signal to the AGI when the call hangs up? |
| 22:50.45 | fgwaller | cant you ceat... ring the extensions simultan.. |
| 22:50.59 | *** join/#asterisk mrdigital (n=wild@pool-72-81-77-174.phlapa.east.verizon.net) |
| 22:51.02 | mrdigital | man this is weird |
| 22:51.05 | syzygyBSD | I have a bunch of defunct processes because the app isn't closing quick enough |
| 22:51.18 | adorah | it is not only a ringing question |
| 22:51.43 | adorah | they work face to face and want it that way.. |
| 22:52.40 | adorah | Is there any booster I can put on the cable to increase the volume for the cordless? |
| 22:53.57 | adorah | In the old days when I had a 1+1 line 12v/6v I used to add a 6v battery to the slave line so faxes could work there..any stupid low tech solution for that to solve it? |
| 22:56.55 | sumasuma | anyone is having g729.c file ? |
| 22:57.11 | sumasuma | my hard disk got screpped |
| 22:57.19 | sumasuma | *got scrapped |
| 22:57.34 | sumasuma | codec_g729.c file |
| 22:57.42 | file | ...no |
| 22:58.00 | mrdigital | file can you help me with a ata problem |
| 22:58.01 | sumasuma | don we need this to use the g729 codec ? |
| 22:58.34 | file | if you want to use the G729 legally, then you have to pay for licenses from Digium and you get a binary module for it |
| 22:58.41 | file | mrdigital, if you ask a question I may answer |
| 22:59.26 | mrdigital | ok |
| 22:59.53 | mrdigital | ATA Device i can call it from softphone.. the ata device rings a analog phone in the FXS Por |
| 23:00.09 | mrdigital | t when i pick up the phone plugged into the ATA Box the softphone keeps ringing |
| 23:00.15 | mrdigital | when i hang the analog phone up |
| 23:00.36 | mrdigital | the softphone plays the person at that extenstion #### is not available please leave a voice message |
| 23:01.40 | file | have you gotten sip debug up on a pastebin? |
| 23:01.48 | mrdigital | sip debug? |
| 23:01.53 | mrdigital | where do i grab that again |
| 23:02.10 | file | you type sip debug on the Asterisk CLI, do your stuff like above, paste it into a pastebin... |
| 23:02.15 | Zodiacal | anyone know if theres a way to get line appearance labels to say the callerid info? |
| 23:02.20 | Zodiacal | on a polycom soundpoint 601 |
| 23:05.41 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.72.107) |
| 23:08.20 | syzygyBSD | k, using deadagi to process the entire agi script made it so it didnt' go defunct |
| 23:09.56 | *** part/#asterisk CANO-1982 (n=alejandr@190.48.72.107) |
| 23:11.13 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 23:11.17 | generalhan | whats up all ! ? |
| 23:11.36 | CrashHD | what format should my music on hold files be in? |
| 23:11.48 | CrashHD | I'm trying pcm 8,000 16bit |
| 23:12.00 | CrashHD | filename is (something.wav) |
| 23:12.02 | CrashHD | will that work ok? |
| 23:13.18 | generalhan | CrashHD: all my MoH files are MP3 |
| 23:13.38 | fgwaller | i have everything in ulaw |
| 23:13.38 | generalhan | placed in the mohmp3 folder |
| 23:13.45 | CrashHD | ya I have mp3's setup right now |
| 23:13.52 | CrashHD | but I want to go native |
| 23:13.58 | CrashHD | think it will sound better |
| 23:14.14 | generalhan | i see .. sorry must have been said right before i came in ! |
| 23:14.20 | generalhan | i dont know anything about native |
| 23:14.21 | fgwaller | solved some MOH issues for me as well... switching all to raw ulaw |
| 23:14.57 | CrashHD | raw ulaw? |
| 23:14.57 | generalhan | I need to know how to pull all the phone numbers sitting in a meet-me room is there anyway to do that ? |
| 23:15.06 | CrashHD | fgwaller: how did you do that in windows? |
| 23:15.20 | fgwaller | on the linux box ;-) |
| 23:16.40 | fgwaller | mpg123 -s --mono -r 8000 -f 4096 -b 10000 ${BASENAME}.mp3 | sox $2 $3 -t raw -r 8000 -s -w -c 1 - -r 8000 -c 1 -w ../mohraw/${BASENAME}.raw |
| 23:16.52 | CrashHD | what packages did you need installed? |
| 23:17.00 | mrdigital | damn now i cant get the box to register with asterisk |
| 23:17.08 | fgwaller | mpg123 & sox |
| 23:17.30 | fgwaller | this will take any mp3 and convert it in raw asterisk format |
| 23:18.29 | Qwell[] | sox should be able to convert mp3 by itself.. |
| 23:18.33 | fgwaller | do you have the right file format for the raw moh .conf |
| 23:19.15 | fgwaller | Qwell: I know but for whatever reason it did not work right... this does work for me... |
| 23:19.27 | *** join/#asterisk Kast (n=Kast@CPE000103870ecf-CM0012c90f3ddc.cpe.net.cable.rogers.com) |
| 23:19.36 | generalhan | Qwell[]: hows it going man ... long time no see ... i have been so busy with other things lately that i havent had much time to be working on Asterisk |
| 23:19.47 | CrashHD | what would be the downsides of using the pcm 8,000 16bit .wav ? it seems to work ok |
| 23:19.53 | fgwaller | I had sox compiled with mp3 and it would break... might be a AMD64 issue |
| 23:20.06 | Lyfe | anyone know why a call from a sip phone (grandstream gxp2000) to an iax softphone (idefisk, jakeniax, loudhush) would go through periods where it sounds fine, and go through periods where it sounds to me (on the iax phone) like the other person is talking through a robot speech synthesizer? |
| 23:20.45 | fgwaller | :q |
| 23:20.53 | fgwaller | sorry |
| 23:20.55 | generalhan | im trying to write a script that will run on my conference room PC to check with asterisk to find out which numbers are in the Meet-me room ... i just need to know how to get that info from * so i can get it from the script .. any ideas ?? |
| 23:21.00 | Kast | Q: I have an extension at my work, and i use a softphone at my house both connected to the same server... I noticed that my phone at work continues to login, keeping my softphone from working, unless i restart it... How can i deny a useragent login |
| 23:23.57 | file | Kast, can't... so don't use the same account for both |
| 23:24.32 | Kast | i can't ip restrict? |
| 23:24.38 | file | creative idea |
| 23:25.21 | file | permit and deny lines might work, try it |
| 23:25.24 | Kast | I'm assuming that's what host= is for in the sip_phones.include ?? |
| 23:25.56 | file | no, that allows you to explicitly specify the IP address in a peer entry |
| 23:26.03 | file | deny=0.0.0.0/0.0.0.0 |
| 23:26.11 | file | permit=192.168.2.0/255.255.255.0 |
| 23:26.37 | Kast | so if i deny= in my .include it will keep my work phone from logging in |
| 23:27.03 | file | that will only allow IPs in that range... but never done that to a peer entry for someone trying to register |
| 23:27.03 | file | try and see |
| 23:27.28 | Kast | ok |
| 23:28.10 | mrdigital | file im getting unauthorized when they ata box tries to register with asterisk but all the information is properlly entered |
| 23:28.24 | file | ok. |
| 23:28.36 | file | I can tell you nothing until I see a sip debug and sip.conf entry for it |
| 23:32.02 | mrdigital | http://pastebin.ca/100265 |
| 23:32.25 | litage | are you violating the GPL (or any other license) if you sell a computer with Asterisk and a G.729 license installed? |
| 23:32.40 | file | that's a subscription, not a register |
| 23:33.03 | mrdigital | huh |
| 23:33.18 | mrdigital | how do i get it so it registers |
| 23:33.20 | *** join/#asterisk niteowldave (n=dave@202.171.182.120) |
| 23:33.27 | file | I'm not a walking manual for ATAs |
| 23:33.47 | file | although your device is stupid |
| 23:33.57 | Delta239 | hey what can i do to have my specific sip trunk to go to a specific queue? |
| 23:34.06 | file | it says it's IP is 0.0.1.102 |
| 23:34.08 | l-fy | litage > which g729? |
| 23:34.25 | file | put nat=yes for it |
| 23:34.27 | litage | l-fy: Digium G.729 codec license |
| 23:34.30 | l-fy | no |
| 23:34.35 | file | mrdigital, that was for you - put nat=yes in it's entry |
| 23:34.46 | mrdigital | i dont know where 0.0.1.102 is coming from |
| 23:34.52 | mrdigital | nat=yes where |
| 23:34.59 | file | in it's entry in sip.conf |
| 23:35.08 | litage | l-fy: "no" as in "the GPL is not violated by selling an Asterisk box with a G.729 license installed"? |
| 23:35.23 | l-fy | you don't violate the gpl because digium owns the copyright of asterisk since is require every developer to allow them to make a proprietary license, so they gave an exception to gpl for g729 |
| 23:35.37 | l-fy | asterisk license is not fully gpl |
| 23:35.49 | l-fy | so you don't violate asterisk license |
| 23:36.06 | litage | l-fy: really? which parts of asterisk aren't GPL'd? |
| 23:36.11 | Qwell[] | I would ask Digium, personally.. |
| 23:36.15 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:36.30 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66) |
| 23:36.31 | Qwell[] | They *do* own the copyright on the g729 module, and distributing it may be a no-no |
| 23:36.33 | mrdigital | File thanks! that got it registered! now 2nd problem is the dialing |
| 23:36.41 | l-fy | hi dlynes_laptop |
| 23:36.44 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:36.44 | l-fy | hi RoyK |
| 23:36.49 | Qwell[] | even though it is free to download |
| 23:37.00 | niteowldave | file: Hi, are you able to help me today with the t38 negotiatoin problem that exists when the ATA has the T38 in the initial SDP invite. |
| 23:37.02 | Lyfe | anyone have any suggestions what would cause the speech in one direction of a conversation between a sip hardphone and an iax softphone to (from the sip to iax phones) to sound robotic, and a bit garbled? |
| 23:37.15 | l-fy | litage > the asterisk license is gpl with some exceptions, and one of them is g729 |
| 23:37.26 | dlynes_laptop | l-fy, good morning! |
| 23:37.32 | file | niteowldave, I don't know of any way to fix that currently |
| 23:37.37 | l-fy | since they have rights over every piece of software is in mainbranch they can do that |
| 23:37.42 | litage | thanks l-fy and Qwell[]. i'm going to email Digium and ask them, just to be safe |
| 23:37.57 | l-fy | litage > you are safe |
| 23:38.10 | file | better to be safe then sorry |
| 23:38.12 | Qwell[] | l-fy: I would have to disagree... |
| 23:38.30 | Qwell[] | unless it's explicitly stated that the g729 module can be redistributed |
| 23:39.14 | Qwell[] | That would be like saying "Can I redistribute a copy of windows with asterisk?" The answer, obviously, is no |
| 23:39.18 | lesouvage | I'm making a voip connection over a flat fee UMTS connection with my Nokia E60. There is a delay of seconds. Is this because of technical reasons or is this because the telco delays the connection to avoid VOIP connections? |
| 23:39.25 | niteowldave | file: It seems that asterisk does not send the invite to the b party unless there is rtp, it sees the t38 and sets up the joint capability for t38 but does not use this to create a channel |
| 23:39.45 | niteowldave | file: without a channel we cannot create a call |
| 23:39.49 | file | niteowldave, the core wants audio... |
| 23:40.08 | l-fy | Qwell > but g729 and asterisk are digium products |
| 23:40.15 | l-fy | well, only the asterisk mainbranch |
| 23:40.21 | Sedorox | g729 is licensed |
| 23:40.30 | Qwell[] | l-fy: okay, so, can I redistribute Windows with, say...some open source Microsoft product? |
| 23:40.33 | Qwell[] | The answer is still no |
| 23:40.35 | Sedorox | digium owns rights to re-license it |
| 23:40.39 | l-fy | yes you can |
| 23:40.44 | l-fy | because asterisk license is not gpl |
| 23:40.49 | l-fy | is gpl with exceptions |
| 23:41.01 | l-fy | including one for openh323 since openh323 is mpl |
| 23:41.04 | RoyK | hi, l-fy |
| 23:41.09 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 23:41.45 | niteowldave | file; should I raise this as a bug in the bug tracker |
| 23:42.38 | file | you can, doesn't mean it'll be fixed soon... |
| 23:43.05 | niteowldave | file: ok, should I add it to 5090 |
| 23:43.29 | *** join/#asterisk anthm (n=anthm@h4608f3b2.area4.spcsdns.net) |
| 23:43.30 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:43.45 | dlynes_laptop | Is there any known memory leaks associated with either 1.2.7.1 or 1.2.9.1? |
| 23:44.50 | dlynes_laptop | I'm experiencing this memory leak on a couple of systems running 1.2.9.1 with Sangoma A200's with echo cancellers, and on another system running 1.2.7.1 with an X100P |
| 23:45.01 | mrdigital | file : http://pastebin.ca/100281 |
| 23:45.06 | mrdigital | according to debug |
| 23:45.08 | dlynes_laptop | They're all running SIP phones, and IAX2 trunking |
| 23:45.20 | mrdigital | asterisk sends the call to the ata device it senses the phone picked up |
| 23:45.35 | bkw_ | it senses? |
| 23:45.44 | mrdigital | yeah it shows the phone was answered |
| 23:45.45 | bkw_ | must be pretty powerful software there :P it has res_cleo? |
| 23:45.48 | bkw_ | haha |
| 23:45.53 | bkw_ | just giving you a hard time :P |
| 23:45.57 | mrdigital | but the softphone just keeps rining |
| 23:46.01 | mrdigital | when i hang the phone up |
| 23:46.12 | mrdigital | the softphone says sorry no one is available to take you call |
| 23:46.17 | file | I don't see in that sip debug that it was answered |
| 23:46.29 | mrdigital | its like it rings the phone but cant connect the call |
| 23:46.35 | mrdigital | see after is says sip/001 is ringing |
| 23:46.41 | mrdigital | then read from 192.168.1.105 |
| 23:46.58 | mrdigital | read from 192.168.1.105 happend when i picked it up |
| 23:47.04 | file | there's no packet to indicate that it was answered |
| 23:47.13 | mrdigital | lemme try 2 softphones |
| 23:47.16 | mrdigital | brb |
| 23:47.16 | file | looks like the SIP stack on this ATA is really really screwed up |
| 23:47.39 | file | or else it's misconfigured badly |
| 23:48.03 | mrdigital | i think its misconfigured |
| 23:49.05 | mrdigital | like when i dial 100 from the phone hooked up to the ata box i get a busy singal |
| 23:49.20 | mrdigital | signal wont dial anything after 2 digits |
| 23:49.42 | Lyfe | anyone have any suggestions for me as to why sound in one direction of a call would be garbled (robotic-sounding) for a portion of the time during a call? |
| 23:50.55 | Lyfe | (and how i might go about fixing it?) |
| 23:54.39 | mrdigital | file: i did xlite-xlite |
| 23:54.42 | mrdigital | and it bridged |
| 23:54.57 | mrdigital | so somereaason ata isnt bridging |
| 23:55.41 | Dovid | hi |
| 23:55.47 | Dovid | anyone know if the snom has QOS ? |
| 23:57.14 | Dovid | ? |
| 23:57.52 | l-fy | Dovid > for sure |
| 23:58.53 | Dovid | l-fy: |
| 23:58.58 | Dovid | l-fy: not working for me |
| 23:59.15 | Dovid | is it on by default ? |
| 23:59.15 | rowter | <PROTECTED> |
| 23:59.23 | Dovid | i have the PC going thru it and when i download big files it goes to shit |
| 23:59.33 | Dovid | do i have to set the TOS setting ? |
| 23:59.43 | l-fy | yes |
| 23:59.59 | Dovid | l-fy: what do i set it to ? now its at 160 |