| 00:05.15 | tomcontr3 | Im having a little problem with asterisk ports I think |
| 00:05.51 | tomcontr3 | I have a friend that can register with SIP protocol from outside my network, but when I call him non of us can hear each other |
| 00:06.06 | tomcontr3 | althought we can answere the call |
| 00:11.25 | *** part/#asterisk jebba (n=jebba@220-179-89-200.fibertel.com.ar) |
| 00:12.35 | *** join/#asterisk sysreq (n=sysreq@bas9-montrealak-1128746697.dsl.bell.ca) |
| 00:15.02 | lirakis | tomcontr3: ~sipnat |
| 00:15.09 | lirakis | ~sipnat |
| 00:15.09 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:17.25 | lirakis | Jan-pp: in sip.conf .. type=peer |
| 00:17.47 | Jan-pp | ok, thx |
| 00:18.37 | Jan-pp | so i should make a section and use register => user:pass@sipgate.de/sectionname or just put type=peer into [general] |
| 00:18.39 | Jan-pp | ? |
| 00:19.39 | tomcontr3 | yeap, I did that |
| 00:19.54 | tomcontr3 | maybe with no sipnat |
| 00:22.40 | lirakis | Jan-pp: you most likely need the register line for your sip provider. you should set type=peer on a per sip user basis |
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| 00:23.31 | Jan-pp | lirakis: when i use the register line there is already a user name (i will be using only one sip connection) |
| 00:24.25 | lirakis | Jan-pp: the register line does not create a callable sip peer |
| 00:24.43 | lirakis | Jan-pp: it only auth's you with your provider so they know where to send your calls |
| 00:24.55 | [TK]D-Fender | ~sipregister |
| 00:24.56 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 00:25.07 | Jan-pp | ah, ok, thx |
| 00:25.33 | [TK]D-Fender | in many cases even INCOMING calls are auth'd. it depends. |
| 00:26.21 | tomcontr3 | its, wird, becaue he gets my calls and a get his, but there is now audio |
| 00:26.52 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 00:27.07 | lirakis | [TK]D-Fender: so what was that about with bkruse? .. i ask b/c i am really curious about the availability of the openmoko phone |
| 00:27.15 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 00:27.50 | tomcontr3 | http://pastebin.ca/654020 |
| 00:28.03 | [TK]D-Fender | lirakis, P0 is not for public consumption and he's spearheading an IAX client for it. |
| 00:29.03 | [TK]D-Fender | tomcontr3, "now", or "no"> |
| 00:29.04 | [TK]D-Fender | ? |
| 00:29.32 | lirakis | [TK]D-Fender: thats awsome... i cant wait to get rid of this BB... |
| 00:29.37 | [TK]D-Fender | tomcontr3, go check all your port forwarding and NAT settings as per the guide. and also note that : |
| 00:29.39 | [TK]D-Fender | ~freepbx |
| 00:29.40 | jbot | somebody said freepbx was unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 00:30.05 | [TK]D-Fender | ok, I'm outta here... later all |
| 00:30.14 | lirakis | l8r |
| 00:30.34 | Jan-pp | ok, thx, i am registered now! |
| 00:30.41 | lirakis | ;) |
| 00:31.19 | Jan-pp | still no incoming calls |
| 00:31.25 | lirakis | @#$! yeah! i think i finally got my homebrew calling card app working! |
| 00:33.38 | lirakis | Jan-pp: are you seeing them hit your cli? |
| 00:36.57 | Jan-pp | http://pastebin.ca/654024 |
| 00:37.02 | Jan-pp | this is my sip.conf |
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| 00:37.40 | Jan-pp | when I call i get a recording "this person is currently not reachable" or something like that |
| 00:37.43 | bbdecompress | i have a live test question |
| 00:37.55 | bbdecompress | my d channels are not coming up |
| 00:38.08 | bbdecompress | yet my lights are gree onin my zttool |
| 00:38.20 | bbdecompress | green in my zttool |
| 00:38.36 | bbdecompress | they can see me when i use loopback |
| 00:38.50 | bbdecompress | what do i need to check for my d channels? |
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| 00:39.09 | Jan-pp | but YES, i get some stuff on the cli! |
| 00:39.42 | Jan-pp | this means that my sip.conf is probably right, doesnt it? |
| 00:39.45 | lirakis | Jan-pp: have you configured extensions.conf? |
| 00:40.04 | Jan-pp | no, at least not correctly |
| 00:40.08 | lirakis | Jan-pp: you need to send the call some where and tell it to ring the extension you created in sip.conf |
| 00:40.37 | Jan-pp | what ext did i "create" in sip.conf? here it is again http://pastebin.ca/654024 |
| 00:40.53 | Jan-pp | sipin? |
| 00:42.11 | lirakis | <PROTECTED> |
| 00:42.57 | Jan-pp | thx |
| 00:43.57 | lirakis | Jan-pp: http://pastebin.ca/654030 |
| 00:44.36 | bbdecompress | [Aug 11 20:41:57] WARNING[7005]: channel.c:3172 ast_request: No translator path exists for channel type Zap (native 76) to 256 |
| 00:44.36 | bbdecompress | [Aug 11 20:41:57] WARNING[7005]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) |
| 00:44.36 | bbdecompress | <PROTECTED> |
| 00:44.36 | bbdecompress | <PROTECTED> |
| 00:44.36 | bbdecompress | sound farmiliar? |
| 00:44.45 | lirakis | Jan-pp: that isnt a whole extensions.conf.. its just the context you defined for sipgate directing the call |
| 00:44.56 | lirakis | bbdecompress: .. i cant believe you didnt get kicked for posting like that |
| 00:45.07 | bbdecompress | all 3 lines? |
| 00:45.36 | lirakis | try 6 lines.. then 5 |
| 00:45.50 | Jan-pp | do i need anything else in the extensions.conf (general context or something?) of course except more lines for the incoming call (wait, pickup, play...) |
| 00:46.22 | lirakis | Jan-pp: ... i mean.. for just recieving an inbound call from a sip did.. thats what you need |
| 00:46.30 | lirakis | Jan-pp: you dont need wait .. or any of that... |
| 00:46.56 | lirakis | Jan-pp: obvioulsy.. you need to change yourdidhere to whatever sipgate sends to you |
| 00:47.04 | lirakis | Jan-pp: or to a pattern .. |
| 00:47.20 | Jan-pp | I do not want to forward the call to any phone, i want asterisk to play a message |
| 00:47.47 | lirakis | <PROTECTED> |
| 00:49.32 | lirakis | Jan-pp: http://pastebin.ca/654032 |
| 00:50.52 | Jan-pp | is 9654085 my id? or is it sipin? or can i make it match anything? |
| 00:51.09 | Jan-pp | or is my id my phone number? how to find out? |
| 00:51.18 | lirakis | <PROTECTED> |
| 00:51.28 | lirakis | <PROTECTED> |
| 00:51.42 | lirakis | <PROTECTED> |
| 00:51.58 | Jan-pp | ok, starting wireshark... |
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| 00:54.52 | rhiliam | Is there something special that you need to do to bridge a call from an FXS channel to an FXO channel? |
| 00:55.00 | lirakis | <PROTECTED> |
| 00:55.22 | Jan-pp | ok, thx, wireshark was kind of messed up |
| 00:55.44 | rhiliam | I have a phone connected to an FXS channel and internal SIP/IAX and incoming ZAP calls make it there fine, but when I pick up the handset and try to dial out - I get a busy tone. |
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| 00:58.50 | Jan-pp | "To: <sip:0049xxxxxxxxxx@sipgate.de>" is what i get, what should i insert - only the number 0049xxxxxxxxx or more? |
| 01:00.05 | Jan-pp | that did not work. isnt there a way to set a "catch-all"? |
| 01:00.26 | lirakis | Jan-pp: you can pattern match it ... exten => _0049XXXXXXXXXX |
| 01:00.36 | lirakis | .. make sure thats the right number of x's |
| 01:00.40 | lirakis | ~pattern |
| 01:00.57 | lirakis | hrm |
| 01:00.59 | lirakis | http://www.voip-info.org/wiki-Asterisk+Dialplan+Patterns |
| 01:02.08 | lirakis | Jan-pp: i guess exten => 0049.,s,1,Answer() |
| 01:02.10 | lirakis | is your best bet |
| 01:02.26 | Jan-pp | why the additional "s,"? |
| 01:03.28 | lirakis | Jan-pp: . .oh sorry about that :p |
| 01:03.29 | Jan-pp | but i inserted the complete phone number and it didnt work (the To-field did not contin any x, i just censored out my phone number) |
| 01:03.41 | *** join/#asterisk daveburr (i=Miranda@208.254.183.84) |
| 01:03.44 | lirakis | Jan-pp: you could just do exten => s,1,Answer() |
| 01:03.55 | lirakis | Jan-pp: that would answer ANYTHING that goes to that context |
| 01:04.04 | *** part/#asterisk daveburr (i=Miranda@208.254.183.84) |
| 01:04.10 | Jan-pp | great |
| 01:04.42 | Jan-pp | but still no luck |
| 01:04.52 | lirakis | Jan-pp: cli and pastebin |
| 01:04.59 | Jan-pp | ok |
| 01:05.49 | Jan-pp | what settings? (verbosity level, shoud i enable sip debug?) |
| 01:06.31 | lirakis | Jan-pp: do 10 verbosity.. no debug now |
| 01:07.45 | Jan-pp | http://pastebin.ca/654038 |
| 01:08.28 | lirakis | Jan-pp: ... thats a paste of an incoming call attempt? |
| 01:08.34 | Jan-pp | yes |
| 01:09.13 | Jan-pp | and the text appeared exactly as i got the recording "this person is currently unreachable" (it has a different wording) |
| 01:09.52 | lirakis | Jan-pp: tyep "sip no debug" |
| 01:09.54 | lirakis | on clie |
| 01:09.56 | lirakis | *cli |
| 01:10.13 | tomcontr3 | I havent been able to solve my NAT problem |
| 01:10.22 | tomcontr3 | anyone could give a little hand? |
| 01:10.22 | Jan-pp | SIP Debugging Disabled |
| 01:10.28 | lirakis | and try again.. i dont care what shows up after the call comes in.. i want to see the setup. |
| 01:10.40 | lirakis | so .. paste from when the call hits |
| 01:12.32 | Jan-pp | http://pastebin.ca/654043 |
| 01:14.09 | lirakis | Jan-pp: yeah thats not even hitting your dialplan |
| 01:14.35 | Jan-pp | ok. where is the problem then (posting configs again, mom plz) |
| 01:15.18 | lirakis | Jan-pp: you should see things more like this: http://pastebin.ca/654044 |
| 01:15.39 | lirakis | Jan-pp: is it a womans voice? ... im trying to determine if its getting to your * box even |
| 01:16.18 | Jan-pp | it is a voice from the phone network i think. but the text i pasted shows up only when a call comes in |
| 01:16.30 | lirakis | Jan-pp: that cli .. looks totally differnt that what is expected ... so i am unclear. A lot of times i can tell now whether its the asterisk recording thats playing back.. or the phone companies |
| 01:17.27 | Jan-pp | http://pastebin.ca/654046 |
| 01:17.45 | Jan-pp | well, the recording includes german text. that should be clear enough... |
| 01:18.37 | lirakis | ha ha SayAlpha doesnt speak like .. text to speach .. it sais the letters individually |
| 01:18.55 | lirakis | ... wow.. my typing is starting to suck .. i need to stop soon |
| 01:18.56 | Jan-pp | i know. so it would make a loooooooong text |
| 01:19.41 | Jan-pp | any idea if there is something wrong with the config? (what you see are the complete files, so maybe something is missing) |
| 01:19.49 | lirakis | Jan-pp: .. this probably isnt the problem... but you dont have () on the ends of answer and hangup |
| 01:20.20 | Jan-pp | fixed |
| 01:20.40 | lirakis | asterisk -rx reload |
| 01:21.02 | Jan-pp | i did /etc/init.d/asterisk restart is that ok? |
| 01:21.32 | lirakis | Jan-pp: you can do that... but you dont need to .. reload is faster.. it just reloads the configs.. not restart asterisk... you can reload without dropping live calls |
| 01:21.39 | Jan-pp | thx |
| 01:21.40 | lirakis | .. you can run reload from the cli too ;) |
| 01:21.55 | lirakis | also .. sayalpha stuff needs quotes |
| 01:22.09 | lirakis | im not sure if that would mess it up either |
| 01:22.30 | Jan-pp | some wiki or so showed it without quotes |
| 01:22.51 | lirakis | Jan-pp: (shrug) .. i dont know if it would or not.. but when your having problems.. better to be safe than sorry |
| 01:23.18 | lirakis | Jan-pp: you might try replacing sayalpha with Playback(tt-monkeys) just for testing |
| 01:24.24 | Jan-pp | <PROTECTED> |
| 01:24.36 | lirakis | yeah |
| 01:24.45 | lirakis | possibly |
| 01:24.49 | lirakis | what are you running ? |
| 01:25.13 | Jan-pp | ??? that was while reload from cli with very high verbosity (12 or so) |
| 01:25.37 | lirakis | Jan-pp: i meant what distro are you running? |
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| 01:26.11 | lirakis | Jan-pp: do you have a fire wall enabled? ... try ip-tables --flush |
| 01:26.21 | Jan-pp | ubuntu server 7.04 with a manually installed 386 kernel inside a vmware (because of broken hw) |
| 01:26.40 | lirakis | Jan-pp: hmm okay... |
| 01:27.13 | Jan-pp | no firewall, going to doublecheck the router |
| 01:27.19 | Jan-pp | what about the acl? |
| 01:27.28 | tomcontr3 | does anyone knows how to use a stun server with asteresik? |
| 01:27.44 | lirakis | Jan-pp: ..i got to get away from the comp... im dyin.. at it since 6 last night lol |
| 01:28.15 | Jan-pp | ok, thx for the help, gues i will go to bed to (its 3:30 AM here...) |
| 01:29.28 | lirakis | Jan-pp: yeah.. better luck when you have a clearer head |
| 01:29.30 | lirakis | l8r |
| 01:29.37 | Jan-pp | l8r |
| 01:29.43 | *** part/#asterisk Jan-pp (n=Jan@frnk-590c421a.pool.einsundeins.de) |
| 01:30.26 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 01:37.35 | tomcontr3 | how can I use a public STUN server? |
| 01:37.48 | tomcontr3 | do I need to register somewhere? |
| 01:44.56 | tomcontr3 | anyone? |
| 01:46.08 | Nivex | afaik a public STUN server is just that. |
| 01:46.35 | Nivex | should answer all comeres |
| 01:46.37 | Nivex | comers |
| 01:46.39 | tomcontr3 | but, how do I use it with my asterisk server? |
| 01:46.46 | Nivex | that I don't know |
| 01:47.23 | tomcontr3 | the problem is that I have an asterisk server behind a nat |
| 01:47.32 | tomcontr3 | and a client that is calling me from outsite |
| 01:47.49 | tomcontr3 | the thing is that I can get those calls, but I cant hear anything |
| 01:48.11 | Nivex | I gather from this that there is no STUN in asterisk: http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+STUN |
| 01:48.37 | Nivex | ah, you may need to look at the nat= directive in your sip.conf for that client |
| 01:49.18 | tomcontr3 | its set as nat=yes |
| 01:50.04 | Nivex | do you have externip= or externhost= set? |
| 01:50.30 | tomcontr3 | let me check |
| 01:50.52 | tomcontr3 | I have |
| 01:50.53 | tomcontr3 | bindport = 5060 |
| 01:50.58 | tomcontr3 | bindaddr = x.x.x.x |
| 01:51.04 | tomcontr3 | localnet = 192.168.1.0/255.255.255.0 |
| 01:51.05 | tomcontr3 | qualify=xxx |
| 01:51.12 | tomcontr3 | and of course the codecs |
| 01:51.42 | Nivex | if you don't have one of those two set, the nat code doesn't know what IP to tell the other end |
| 01:51.55 | Nivex | and then of course you have to have ports open on your firewall |
| 01:52.08 | tomcontr3 | mmm |
| 01:52.14 | Nivex | see /etc/asterisk/rtp.conf for those ports |
| 01:52.16 | tomcontr3 | so what should I add or change here? |
| 01:52.20 | Nivex | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:52.28 | tomcontr3 | I did map those ports 10000:20000 |
| 01:53.10 | Nivex | you'll want to make sure externip=x.x.x.x is set to your actual public IP |
| 01:53.37 | Nivex | if you have DynDNS or some such pointing at your IP, you can set externhost=somehost.dyndns.org |
| 01:54.09 | tomcontr3 | and what about bindaddr |
| 01:54.14 | Nivex | what about it? |
| 01:54.44 | tomcontr3 | there I should put my local ip? |
| 01:55.12 | Nivex | I have mine set to 0.0.0.0. That tells it to bind on all available interfaces. |
| 01:55.21 | tomcontr3 | ok |
| 01:55.45 | tomcontr3 | qualify= to yes? |
| 01:56.11 | Nivex | irrelevant |
| 01:56.55 | tomcontr3 | and should I set NAT=yes there? or just in the extensions configs? |
| 01:57.23 | Nivex | I set nat=yes only for the specific peer that requires it |
| 01:58.06 | tomcontr3 | ok |
| 01:58.11 | tomcontr3 | I will tryit again now |
| 02:04.53 | tomcontr3 | it worked |
| 02:04.55 | tomcontr3 | thanks a lot |
| 02:23.55 | hmmhesays | everybody's working for the weekende |
| 02:30.47 | rudholm | it seems odd that "SIT" isn't in indications.conf |
| 02:31.34 | rudholm | is there some easy way to send SIT tones other than specifying the frequencies/timings in Playtones() ? |
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| 03:04.55 | tomcontr3 | im having a problem with the CDR |
| 03:05.00 | tomcontr3 | the database is blak |
| 03:05.04 | tomcontr3 | blank |
| 03:05.25 | tomcontr3 | I have checked that the User and Password are correct |
| 03:06.02 | Corydon76-home | What backend are you using? |
| 03:06.21 | tomcontr3 | 1,2,24 |
| 03:06.34 | Corydon76-home | No, what database? |
| 03:06.42 | tomcontr3 | ohh mysql 5.1 |
| 03:06.56 | Corydon76-home | Are you using cdr_odbc or cdr_addon_mysql? |
| 03:07.02 | tomcontr3 | sorry mysql 5.0 |
| 03:07.21 | tomcontr3 | I think it is th cdr_addon_mysql |
| 03:07.31 | Corydon76-home | Please check |
| 03:07.32 | tomcontr3 | I use cdr_mysql.conf |
| 03:07.39 | *** join/#asterisk thesndguru (n=Nathan@202.61.168.243) |
| 03:07.42 | Corydon76-home | Is the module loaded? |
| 03:07.52 | tomcontr3 | yes |
| 03:08.10 | tomcontr3 | I mean I thinks so, becaue asterisk didnt complain |
| 03:08.40 | tomcontr3 | cdr_custom.so Customizable Comma Separated Values CDR 0 |
| 03:08.57 | tomcontr3 | cdr_csv.so Comma Separated Values CDR Backend 0 |
| 03:09.22 | Corydon76-home | Well, there's your problem |
| 03:09.28 | tomcontr3 | 0? |
| 03:09.40 | Corydon76-home | If you haven't loaded the driver, then you won't get an entry in the database |
| 03:09.59 | tomcontr3 | how do I loadthem? |
| 03:10.15 | Corydon76-home | Well, first you need to get the asterisk-addons source and compile it |
| 03:10.51 | tomcontr3 | I did it |
| 03:11.04 | tomcontr3 | I have those modules on my modules folder |
| 03:11.04 | Corydon76-home | Did you 'make install'? |
| 03:11.09 | tomcontr3 | yep |
| 03:11.16 | Corydon76-home | then 'load cdr_addon_mysql.so' |
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| 03:12.10 | tomcontr3 | mmm thats wird the module does not existe |
| 03:12.27 | Corydon76-home | You probably don't have the mysql headers installed, then |
| 03:13.00 | Corydon76-home | Install the headers, then rebuild asterisk-addons |
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| 03:13.47 | tomcontr3 | ohhh I see |
| 03:16.08 | thesndguru | wondering if someones knows a way of checking for hang up during a dialplan |
| 03:16.21 | thesndguru | i'm getting calls to message bank |
| 03:16.52 | thesndguru | with the hang up tone / calls going to the lines in home that have already hang up |
| 03:17.28 | thesndguru | the stupid telemarketers test if your home calls |
| 03:17.45 | thesndguru | if that makes sence to anyone? |
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| 03:46.30 | twistedolive | Hi. Just wondering if anyone knows of any existing guide on setting up Asterisk on Mac with OS X.... |
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| 03:49.50 | twistedolive | I've compiled and installed Asterisk on my mac. However, it appears that it's got address 0.0.0.0. I'm not sure what to do from there. |
| 03:50.21 | jql | 0.0.0.0 means "all addresses" |
| 03:51.13 | jql | well, in the config at least. :) |
| 03:53.27 | twistedolive | it's got "SIP Listening on 0.0.0.0:5060" and the same for all the other IP addresses except for the ports |
| 03:54.12 | twistedolive | just thought that it should be the network IP address of my mac on there? |
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| 04:07.39 | twistedolive | After installation of Asterisk, do I have to do a "manual setup" for its network address? Or does Asterisk automatically grab the IP address of the active network connection? |
| 04:07.52 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 04:08.00 | twistedolive | greatly appreciated if anyone could give me some pointers |
| 04:08.10 | [TK]D-Fender | twistedolive, the sample configs bind to all open interfaces. |
| 04:09.37 | SwK | twistedolive, look at the sample configs for the protocol you want to use it shows you how to bind to specific interfaces in there |
| 04:13.20 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 04:17.54 | *** part/#asterisk twistedolive (n=twistedo@CPE001310d974eb-CM001371165cd8.cpe.net.cable.rogers.com) |
| 04:22.16 | *** join/#asterisk saftsack (n=saftsack@pD9E07347.dip.t-dialin.net) |
| 04:25.22 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:27.16 | *** join/#asterisk implicit_ (n=implicit@210.16.55.38) |
| 04:53.40 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
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| 04:54.34 | Weezey | I just upgraded my asterisk setup to latest 1.4 and everything works (that I've tested so far anyway) except I can't hear any prompts. |
| 04:55.09 | Weezey | I call in and when I do a tcpdump, I only see packets from my phones, nothing from * back to them. |
| 04:55.09 | Weezey | console says i |
| 04:55.20 | jql | rtp debug on |
| 04:55.27 | jql | check where asterisk thinks its sending them |
| 04:55.31 | Weezey | k |
| 04:58.02 | Weezey | sees the public IP but isn't sending anything back. |
| 04:58.36 | jql | so, no "Sending packet to XXX" scrolling across the screen? |
| 04:58.41 | Weezey | none. |
| 04:58.54 | jql | is there "Received RTP from xxx" at least? |
| 04:58.56 | Weezey | just Got RTP packet from |
| 04:59.00 | jql | yeah, got |
| 04:59.01 | jql | umm... |
| 04:59.03 | Weezey | lots |
| 04:59.10 | jql | yeah, 50 per second. :) |
| 04:59.26 | Weezey | calls work though, just not talking to the * box. |
| 05:00.23 | jql | well, that's not good. I'd have to see a debug trace to follow what's going wrong |
| 05:00.52 | Weezey | how do I build what you need? |
| 05:00.57 | jql | sip debug on, core set verbose 10, core set debug 10 |
| 05:01.07 | jql | then make a call (rtp debug off first) |
| 05:01.24 | Weezey | like a normal call that works? |
| 05:01.34 | Weezey | should I only turn on sip debugging for this peer? there's lots |
| 05:02.02 | jql | yeah, trace the broken one |
| 05:02.28 | Weezey | all phones do this, coming in on iax or sip |
| 05:02.37 | Weezey | didn't try google talk though.. |
| 05:03.10 | jql | iax, too? well, that should be impossible. do me a favor: in asterisk.conf, set internal_timing=yes |
| 05:05.27 | Weezey | restarting... |
| 05:06.34 | Weezey | (that's another issue, if you do: restart when convenient from a connected console (asterisk -r) it seems to lock up while trying to unload something whereas if you do it from the console it works just fine.) |
| 05:07.13 | jql | hmm. I haven't noticed that |
| 05:07.47 | Weezey | could be my setup |
| 05:08.12 | Weezey | I need spandsp and I have sangoma AND digium analog cards in there. |
| 05:08.24 | jql | fun |
| 05:08.27 | Weezey | I'm not worried about that, but I need to hear my prompts |
| 05:08.39 | jql | very true |
| 05:08.50 | Weezey | doesn't seem to have done anything. |
| 05:09.03 | jql | rtp debug still shows one-way packets? |
| 05:09.06 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 05:09.53 | Weezey | yep |
| 05:09.58 | jql | oh boy |
| 05:10.30 | jql | yeah, best thing to do is get a good log saved |
| 05:10.33 | Weezey | let me try that same check from one of the phones which connects to this box via iax |
| 05:12.07 | Weezey | hmm, no rtp at all |
| 05:12.36 | Weezey | tcpdump shows one way iax though. |
| 05:13.05 | Weezey | okay, so what am I logging? |
| 05:13.24 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 05:13.44 | jql | personally, I enable full logging via logging.conf, turn on debug 10, verbose 10, sip debug, and then make a quick test call |
| 05:13.56 | jql | then grab the relevant portion from /var/log/asterisk/full |
| 05:14.28 | jql | err, logger.conf |
| 05:20.17 | Weezey | k, you want this or should I be sending it elsewhere? |
| 05:26.42 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 05:27.49 | Weezey | i need sleep |
| 05:27.55 | Weezey | I'll fight this fight tomorrow |
| 05:27.58 | Weezey | thanks for the help |
| 05:28.41 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 05:32.45 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
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| 05:42.43 | *** part/#asterisk sQuEE` (n=vilya@202-195-235-201.fibertel.com.ar) |
| 05:55.18 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 05:55.48 | asterisknerds | <PROTECTED> |
| 06:24.01 | *** join/#asterisk mtaht4 (n=m@cpe-065-190-150-008.nc.res.rr.com) |
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| 06:42.22 | *** join/#asterisk BigA_ (n=root@schizotx.kdtsolutions.com) |
| 06:43.01 | BigA_ | hello |
| 06:46.17 | LoneShadow | what does this command mean "exten => h,1,SetCIDNum(${CALLERIDNUM:0}) |
| 06:46.18 | LoneShadow | ? |
| 06:46.32 | LoneShadow | Is it resetting the callerid number with 0 ? |
| 06:47.03 | jql | no, that's the dialplan substring syntax |
| 06:47.35 | LoneShadow | so what does it mean ? |
| 06:47.53 | jql | it's actually doing nothing, but ${CALLERIDNUM:3} would strip off the first three numbers |
| 06:48.22 | jql | for instance, $CALLERIDNUM = "123456789", ${CALLERIDNUM:3} == "456789" |
| 06:48.40 | LoneShadow | if I do "NoOp,${CALLERID(num)}" I can see the caller's number |
| 06:48.47 | LoneShadow | but not with CALLERIDNUM |
| 06:48.58 | jql | yeah, calleridnum is old and deprecated |
| 06:49.09 | jql | CALLERID(num) and CALLERID(name) are the new ones |
| 06:49.17 | LoneShadow | ok |
| 06:49.40 | LoneShadow | so if I were to do SetCIDNum(${CALLERID(num)}) |
| 06:49.42 | LoneShadow | that fails |
| 06:49.51 | jql | yeah, that function is old and busted too |
| 06:50.06 | jql | Set(CALLERID(num)=${CALLERID(num)}) |
| 06:50.06 | JT | that is completely defective syntax :P |
| 06:50.10 | jql | would be a noop |
| 06:50.19 | BigA_ | has anyone tried to make a MEGACO phone work? I used the MGCP channel and get lots of errors about "Message must have a verb, an idenitifier, version, and endpoint" on the console when the phones try to register. i don't really know what im doing btw |
| 06:51.04 | LoneShadow | so I dont even need to do Set or SetCIDNUM then |
| 06:51.19 | jql | if it has a value, it has a value |
| 06:55.37 | LoneShadow | also do rules always have to all numbers in sequence. If I were to comment out a line, do I need to renumber the remaining ? |
| 06:56.00 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 06:56.06 | jql | you may want to find a newer reference |
| 06:56.07 | jql | :) |
| 06:56.30 | asterisknerds | <PROTECTED> |
| 06:56.50 | LoneShadow | err |
| 06:56.57 | jql | there is an auto-numbering feature. use n instead of a number, and it will assign one for you, as long as you set the first priority in the extension explicitly to 1 |
| 06:57.31 | LoneShadow | oh nice |
| 06:57.33 | jql | exten => 123,1,NoOp(Doing 123) .... exten => 123,n,DoStuff |
| 06:57.33 | LoneShadow | thanks |
| 06:57.37 | jql | that's what I do |
| 06:57.48 | LoneShadow | sweet, didnt know about that, always renumbered :D |
| 06:57.51 | jql | then you can comment any action line out, as long as you leave the NoOp |
| 07:00.28 | LoneShadow | jql: do you use freepbx ? |
| 07:00.36 | jql | nope |
| 07:00.52 | jql | well, not in production. just keep an eye on it. :) |
| 07:02.28 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:03.15 | LoneShadow | I am trying to copy the callback example from nerdsvittle |
| 07:08.46 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 07:15.03 | BigA_ | can anyone recommend a cheap ip phone to work with asterisk? |
| 07:15.12 | foo | BigA_: Why not use a softphone? |
| 07:15.22 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 07:15.32 | BigA_ | i do, plus an fxs to regular phone.. just wanted to experiment |
| 07:15.40 | foo | ah |
| 07:16.30 | BigA_ | i actually have a couple nec phones that do megaco but i cant figure out how to get em working, or if its even possible |
| 07:22.18 | *** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima) |
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| 07:45.58 | HaMYaI | anyone knows why ChanSpy hasn't been working with pure ZAP channels for a while already |
| 07:46.37 | HaMYaI | it will only work if ZAP is bridged with other channels, SIP for instance |
| 07:53.47 | Daviey | hmm.. in what situation would a sip trunk be registered, not show any output when a call is made - but with sip debug on, does show? |
| 07:54.05 | *** join/#asterisk Uatec (n=uatecuk@77.241.176.36) |
| 07:54.09 | Daviey | "call is made" = call is recieved to * |
| 07:57.42 | tzafrir | HaMYaI, if you add the 't' flag to the dial option, will it work? |
| 07:57.59 | JT | no trunks with sip... |
| 07:59.57 | *** join/#asterisk arctis (i=arctis@63.219.95.219.brf02-home.tm.net.my) |
| 08:00.18 | HaMYaI | tzafrir: the incoming ZAP does use Dial |
| 08:00.28 | *** join/#asterisk af_ (n=getsmart@81-174-8-110.dynamic.ngi.it) |
| 08:01.23 | HaMYaI | tzafrir: I play them the IVR menu for each meetme they want to enter and just wanna use ChanSpy while they are in meetme |
| 08:02.17 | tzafrir | HaMYaI, I asked you if you could try a simple test: add the option 't' to the options you use for Dial there and see if it changes anything |
| 08:03.23 | HaMYaI | tzafrir: I am using Dial (with no t) and it works |
| 08:04.01 | *** join/#asterisk xtr-II (i=01928375@216.19.191.191.novuscom.net) |
| 08:06.06 | tzafrir | But what about ChanSpy? |
| 08:06.09 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 08:07.39 | *** join/#asterisk ecam (n=ecam@bb219-75-109-177.singnet.com.sg) |
| 08:08.47 | ecam | hey, i have this weird problem with my asterisk setup, the itsp will suddenly ignore any sip packets i send to it, even though the itsp can still send incoming calls to me |
| 08:08.56 | HaMYaI | tzafrir: sorry, I probably confused you. I mean the incoming ZAP doesn't use Dial |
| 08:09.58 | ecam | anyone with any idea why this is so? |
| 08:10.11 | HaMYaI | tzafrir: and ChanSpy works for the bridged ZAP <-> SIP channels, that is I use Dial(SIP/..) from ZAP channel |
| 08:10.51 | ecam | and the itsp will suddenly start responding again after quite a while, how long, i dont know |
| 08:11.20 | HaMYaI | tzafrir: but for ZAP, I can only use ZapBarge and ZapScan |
| 08:15.13 | *** join/#asterisk xtr (i=94752345@216.19.191.191.novuscom.net) |
| 08:15.38 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 08:17.29 | ecam | anyone can help? a little annoyed that I can't place outgoing calls |
| 08:17.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:32.37 | *** join/#asterisk Chai_Sangeen (n=Chai_San@84.255.156.228) |
| 08:32.42 | Chai_Sangeen | hello everyone |
| 08:33.03 | BigA_ | hi |
| 08:33.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:36.12 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
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| 08:40.29 | ecam | let me ask again: i have this weird problem with my asterisk setup, the itsp will suddenly ignore any sip packets i send to it, even though the itsp can still send incoming calls to me |
| 08:41.14 | *** part/#asterisk HaMYaI (i=HaMYaI@125-25-197-107.adsl.totbb.net) |
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| 08:55.45 | Chai_Sangeen | i'm running my asterisk server behind a nat. I modified the sip_nat.conf “externhost=ip-address” and it works fine with remote clients with no problems at all. I don't have a static ip, so when i use my domain name in “externhost” i can still register to the server from remote location, but with one way audio problem.. i've been struggling with problem for a very long time, and tried about every solution i found on the net. is there a fi |
| 08:55.45 | Chai_Sangeen | x for this, or a script that automatically modifies the wan ipaddress in externhost once it changes? another strange problem with my working configuration my remote sip client is registered to the server but on port 2050 and the client is configured to use port 5060, but no problems with audio. i'll appreciate it if anyone can help. Here is a link to my sip_nat.conf and some other settings http://paste.ubuntu-nl.org/33424/ |
| 08:56.07 | *** join/#asterisk thansen|laptop (n=thansen@74-36-210-143.dr01.hmdl.id.frontiernet.net) |
| 08:56.39 | Chai_Sangeen | sorry this is the updated link: http://paste.ubuntu-nl.org/33426/ |
| 09:00.24 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
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| 10:30.31 | knarfly | my pots line connection using an X101P has an echo...but my zapata.conf has echocancel=yes...what gives? |
| 10:30.48 | JT | it's a piece of junk |
| 10:31.01 | JT | and software echo cancellation will never be as good as hardware echo cancellation |
| 10:31.17 | knarfly | yes, but the person I call doens't hear the echo...only I hear my own voice echo |
| 10:31.40 | knarfly | JT, what card do you use? |
| 10:31.40 | JT | do you hear echo from the far end? |
| 10:32.07 | tzafrir | knarfly, something to try: http://www.rowetel.com/ucasterisk/oslec |
| 10:32.13 | knarfly | the people I call don't complain of an echo and their voice doesn't echo...only mine |
| 10:32.25 | JT | i generally avoid analogue, and if i did go analogue i'd probably use an ATA or an A200 |
| 10:32.40 | JT | actually i have an analogue channel bank, but it's not hooked up to the PSTN |
| 10:33.15 | JT | what phone are you using? |
| 10:33.29 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 10:33.30 | knarfly | grandstream BT200 |
| 10:33.37 | JT | ... |
| 10:33.49 | JT | that's more important than the fact you're using an X101P |
| 10:33.57 | JT | id' wager that's why you're getting echo |
| 10:34.01 | JT | grandstreams are utter junk |
| 10:34.04 | JT | ~gs |
| 10:34.09 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 10:34.19 | knarfly | yes, but they are cheap and I |
| 10:34.26 | knarfly | yes, but they are cheap and I'm a tightwad |
| 10:35.06 | knarfly | oneday I will spend the bucks and improve my equipment |
| 10:35.55 | JT | only they know how to suck so much to make the phone user's sidetone sound like echo ;) |
| 10:36.45 | knarfly | tzafrir, that looks promising...but I run FreeBSD not Linux...I wonder if it will work on my setup? |
| 10:37.54 | JT_ | no, it won't help. |
| 10:38.01 | JT_ | the problem is internal to your phone |
| 10:38.10 | JT_ | problem is resolved by throwing phone in bin |
| 10:38.18 | JT_ | a softphone is better than a BT200 |
| 11:12.49 | *** join/#asterisk saftsack (n=oliver@p54A7CB1D.dip.t-dialin.net) |
| 11:22.36 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:30.13 | tzafrir | JT_, hmmm... are you sure? A softphone tends to have a big latency, and hence will have echo. Some softphones have non-existing echo cancellers |
| 11:35.38 | *** join/#asterisk saftsack (n=oliver@p54A7CEEC.dip.t-dialin.net) |
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| 12:02.23 | JT | tzafrir: sidetone "echo"... |
| 12:06.09 | JT | that's seriously messed up |
| 12:07.07 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 12:08.01 | apardo | somebody knows wich module i need for detect a hangup from the remote party? i'm building a minimal system, only IP |
| 12:08.14 | apardo | i'm testing with iax |
| 12:13.57 | coppice | most IP phones lack sidetone cancellers. that's really crappy. early IP phones generally had them |
| 12:15.09 | *** join/#asterisk sysreq (n=sysreq@bas9-montrealak-1128746697.dsl.bell.ca) |
| 12:15.34 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 12:16.31 | apardo | the scenario is my_mystem <-> another_asterisk <-> pstn |
| 12:16.49 | apardo | when pstn hangup the channel my system don't detect it |
| 12:18.08 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 12:18.09 | hi365 | hey, im geting this weird thing where asterisk will hangup if i use the p option (in dial) ONLY ON SPECIFIC TRUNKS! |
| 12:18.52 | apardo | this is my modules config: http://rafb.net/p/bn59mn84.html |
| 12:19.42 | apardo | hi365: go to test your suggestion |
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| 12:28.02 | *** join/#asterisk rodo (n=rodo@home.lafrere.net) |
| 12:28.17 | rodo | Hi from France |
| 12:28.26 | *** join/#asterisk saftsack (n=oliver@p54A7FDDA.dip.t-dialin.net) |
| 12:29.35 | *** join/#asterisk d1avlo (n=thi@189.13.193.76) |
| 12:29.51 | xheliox | Anyone ever played with SoftEcho from OctWare? |
| 12:31.05 | d1avlo | i dont call :( |
| 12:31.21 | *** join/#asterisk Whoo (n=whoo@ADSL-TPLUS-15-189.intnet.mu) |
| 12:34.08 | Whoo | Hi all .. |
| 12:36.08 | Whoo | I've configured two sip account ...both are connected (Registred). "sip show peers " show them. |
| 12:36.08 | Whoo | But a call say : "unknown user" |
| 12:36.27 | *** join/#asterisk elixer (n=seanbrig@c-69-251-152-9.hsd1.md.comcast.net) |
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| 12:44.05 | *** join/#asterisk d1avlo (n=thi@189.13.193.76) |
| 12:44.32 | d1avlo | i have problem => http://paste.milk-it.net/545 |
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| 13:25.22 | x86 | morning |
| 13:25.56 | *** join/#asterisk cayorde (n=flexable@87.19.110.186) |
| 13:26.18 | Whoo | I'm looking for a very simple tutorial to use SIP (only) with asterisk ... |
| 13:28.11 | tzafrir | Whoo, SIP to what device? or what soft-phone? |
| 13:28.22 | tzafrir | Take a look at voip-info.org |
| 13:30.15 | Whoo | juste to use ekiga |
| 13:30.28 | Whoo | with some different computer |
| 13:32.05 | Whoo | I understand that I need to configure sip.conf and extension.conf ... use can connect but, can't call together |
| 13:33.06 | tzafrir | make a simple type=friend entry |
| 13:33.11 | tzafrir | in sip.conf |
| 13:33.53 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 13:34.19 | tzafrir | make sure it has host=dynamic |
| 13:34.27 | tzafrir | now run 'sip reload' |
| 13:34.48 | tzafrir | do you see your phone in 'sip show peers' (with its IP address)? |
| 13:35.09 | Whoo | yes |
| 13:35.52 | Whoo | >> http://pastebin.ca/654468 |
| 13:41.53 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:41.53 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:43.14 | Weezey | I'm having one way audio when I connect to any prompt (background/playback). I updated by box from 1.2-trunk to 1.4-trunk yesterday. If I do an rtp debug, I only see packets from the caller, not back from the * box. |
| 13:45.45 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 13:50.20 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:50.31 | Whoo | YES... :) the connection working :) |
| 13:55.21 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 14:01.59 | *** join/#asterisk iGimmick (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 14:03.28 | *** part/#asterisk iGimmick (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 14:10.02 | Whoo | Arf ... I've got a little trouble, there is no sound ... ekiga switching fast between PCMCU/blank... |
| 14:11.24 | tzafrir | first-off, it may be simpler to test a connection from one phone to an echo-test extension (Echo) than between two phones |
| 14:19.35 | *** join/#asterisk CVirus (n=GoD@217.54.232.146) |
| 14:23.40 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 14:30.25 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 14:36.54 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 14:39.15 | *** join/#asterisk Ebola (n=Ebola@host86-144-86-8.range86-144.btcentralplus.com) |
| 14:41.46 | *** join/#asterisk prizark (i=prizark@bb220-255-230-20.singnet.com.sg) |
| 14:50.40 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:50.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:52.12 | hi365 | hello. im my sip DID suddenly started to dissconect when i use the p(privacy) option with dial |
| 14:52.17 | _Krieger_ | must we edit both extensions.ael and .conf to keep them similar? or... |
| 14:52.26 | hi365 | the cli is reporting: app.c: No audio available on SIP/" |
| 14:52.28 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 14:52.50 | hi365 | what could be the cause of this? |
| 14:53.01 | hi365 | my ports are all open. other did's work fine |
| 14:57.36 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) [NETSPLIT VICTIM] |
| 15:09.56 | *** join/#asterisk manopulus (n=manopulu@cable-12-21.cgates.lt) |
| 15:13.32 | manopulus | hello. I have a question, maybe someone can answer here. I building macro, where I want to use variables as array. I have arguments, when calling macro, i.e. macro(dial,counter,Zap/g1/123,,tT,IAX2/mytrunk/12345,30,tT) and so on. Within macro I have to get value from next argument. I have tried: |
| 15:13.32 | manopulus | Set(destination=${ARG$["${counter}"]}), |
| 15:13.32 | manopulus | Set(destination=${ARG$[${counter}]}), |
| 15:13.32 | manopulus | Set(destination=${ARG${counter}}), |
| 15:13.33 | manopulus | and each time I have error: |
| 15:13.35 | manopulus | <PROTECTED> |
| 15:13.37 | manopulus | Aug 12 18:09:27 WARNING[8966]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: |
| 15:13.40 | manopulus | So, what is wrong, any idea how to resolve ? |
| 15:15.51 | hi365 | why dont you just call the next argumnent? |
| 15:16.29 | hi365 | ${ARG1} or perhaps $[${ARG!}+1] |
| 15:16.49 | manopulus | and what about ARG N ? |
| 15:17.06 | manopulus | I have unlimited quantity :) |
| 15:17.31 | hi365 | hmm, are you using goto? |
| 15:17.36 | SwK | <PROTECTED> |
| 15:17.47 | SwK | and do something like ${ARG${ITTERATION} |
| 15:17.53 | SwK | err }} at the end not just 1 |
| 15:18.05 | *** join/#asterisk Tili (n=tili@153.Red-80-38-134.staticIP.rima-tde.net) |
| 15:18.20 | manopulus | $[${ARG!}+${counter}] |
| 15:18.22 | manopulus | like this ? |
| 15:22.17 | SwK | Variables are nestabout so where foobar = 1234, wtf=bar, ${foo${wtf}} == ${foobar} == 1234 |
| 15:23.32 | *** part/#asterisk Whoo (n=whoo@ADSL-TPLUS-15-189.intnet.mu) |
| 15:24.36 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:28.36 | *** join/#asterisk RageMax (n=RageMax@pool-72-77-124-103.pitbpa.east.verizon.net) |
| 15:28.57 | Qwell | Set(destination=${ARG${counter}}) should work just fine.. |
| 15:30.17 | manopulus | now checking for counter value, I did that and get dummy value... but for some reason, I have to use _X, in macro, not s |
| 15:31.01 | *** join/#asterisk shareenergy (n=shareene@195-23-137-26.net.novis.pt) |
| 15:31.20 | shareenergy | can anyone help me setup a default inbound route in asterisk? |
| 15:40.20 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:47.51 | tzafrir | I know that using the pattern "_." is generally not recommended. But is there a pattern that means "any number? |
| 15:48.25 | tzafrir | "_X." assumes that the extension number has more than one digit. And this fails in some cases |
| 15:48.34 | tzafrir | Any clever work around? |
| 15:49.17 | [TK]D-Fender | tzafrir : _X! |
| 15:49.26 | tzafrir | aha! |
| 15:49.48 | tzafrir | Is this good for Asterisk 1.2? |
| 15:49.55 | [TK]D-Fender | i believe so |
| 15:50.06 | tzafrir | ok, /me goes for some reading |
| 15:52.55 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:54.08 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 15:55.20 | *** join/#asterisk ccesario_ (n=QUui@201-0-53-62.dsl.telesp.net.br) |
| 16:07.24 | tzafrir | [TK]D-Fender, works well in 1.2 |
| 16:08.04 | [TK]D-Fender | tzanger, ! means "this or anything longer so long as its unique from all other matches" |
| 16:08.09 | [TK]D-Fender | IIRC |
| 16:08.22 | [TK]D-Fender | I only recently became aware of it myself |
| 16:08.39 | tzafrir | so it has a lower priority than _X. ? |
| 16:10.08 | [TK]D-Fender | I don't even want to THINK about a context with BOTH! |
| 16:10.47 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:19.42 | *** join/#asterisk gardo (n=gardo@121.97.245.10) |
| 16:21.30 | *** join/#asterisk limb_q (n=root@limbique.xs4all.nl) |
| 16:21.32 | limb_q | hi all |
| 16:21.44 | limb_q | whois |
| 16:21.58 | limb_q | hmm... :) running with ircii |
| 16:22.14 | limb_q | can't find a solution on my problem.. |
| 16:22.37 | limb_q | i'm receiving a sip message back from my provider: |
| 16:22.47 | limb_q | SIP/2.0 501 not implemented yet |
| 16:22.49 | limb_q | anyone? |
| 16:23.11 | limb_q | what can it be... where should i look...? :S |
| 16:25.54 | limb_q | tzanger.. |
| 16:27.02 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 16:27.08 | limb_q | hi |
| 16:27.12 | Voicemeup | any reson i dont hear Playing 'vm-login' (language 'fr') |
| 16:27.15 | Voicemeup | its silent |
| 16:27.17 | Voicemeup | anythign else works |
| 16:27.34 | limb_q | idk, soz |
| 16:27.52 | LoneShadow | anyone using sphinx ? |
| 16:27.58 | limb_q | i'm trying to connect my asterisk on my sip provider. :S |
| 16:28.04 | limb_q | i get a SIP/2.0 501 not implemented yet |
| 16:28.30 | limb_q | the're using Cirpack/v4.41 (gw_sip) ??? |
| 16:28.34 | limb_q | ever heard of? |
| 16:29.05 | tzafrir | limb_q, what's the other party? |
| 16:29.09 | tzafrir | ah, ok |
| 16:29.16 | limb_q | hi tza |
| 16:29.18 | limb_q | ty |
| 16:29.32 | tzafrir | never heard of that Cirpack :-( |
| 16:29.50 | limb_q | i'm running deb with |
| 16:30.00 | limb_q | Linux home-pbx 2.6.21-2-486 #1 Wed Jul 11 03:17:09 UTC 2007 i686 GNU/Linux |
| 16:30.20 | limb_q | Asterisk 1.2.13 |
| 16:30.23 | [TK]D-Fender | limb_q, WHEN do you get this message? |
| 16:30.33 | limb_q | i see my pbx sends a msg |
| 16:30.36 | LoneShadow | I wonder if its possible to say "Press 1 or say one" and catch either of those 2 in asterisk |
| 16:31.24 | Voicemeup | hmm so i Answer() Wait(1) VoicemailMain() |
| 16:31.25 | *** join/#asterisk mtaht4 (n=m@cpe-065-190-150-008.nc.res.rr.com) |
| 16:31.25 | LoneShadow | limb_q: does your SIP provider expect you to use a certain sip device ? |
| 16:31.26 | Voicemeup | weird |
| 16:31.37 | Voicemeup | no audio for mailbox name.. but ok for password lol |
| 16:32.38 | limb_q | i searched forums |
| 16:32.52 | [TK]D-Fender | limb_q, again, WHEN do you get this message? |
| 16:33.15 | limb_q | i did sip debug |
| 16:33.36 | limb_q | the first msg my pbx sends is .. can i post it somewhere? |
| 16:33.40 | [TK]D-Fender | limb_q, Yes, thats clearly the only reason you should SEE it, but I asked WHEN. |
| 16:33.48 | [TK]D-Fender | ~pb |
| 16:33.49 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:33.49 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 16:34.06 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:34.17 | limb_q | i post some logging ok? |
| 16:34.36 | *** join/#asterisk decobbb (n=bbradley@66.255.125.146) |
| 16:35.53 | decobbb | does anyboday have any experience hooking * up via 2 PRI lines? |
| 16:36.33 | limb_q | http://paste.debian.net/34480 |
| 16:37.15 | [TK]D-Fender | limb_q, set "qualify=no". its spamming that back because of your attempted keep-alive that they don't support |
| 16:37.40 | limb_q | i'll try that |
| 16:37.41 | limb_q | moment |
| 16:38.46 | limb_q | can't see any sip messages yet.. |
| 16:38.50 | limb_q | i'll try the number |
| 16:39.29 | *** join/#asterisk saftsack (n=oliver@p54A7F20C.dip.t-dialin.net) |
| 16:39.32 | limb_q | i get a message, when i call that number, like i'm offline |
| 16:39.58 | limb_q | "the person your trying to reach, is unavailable" translated |
| 16:40.48 | limb_q | i'll post my sip.conf (without pwds) |
| 16:44.33 | limb_q | http://paste.debian.net/34482 |
| 16:44.55 | limb_q | so u can see, i altered the qualify |
| 16:46.04 | [TK]D-Fender | can reinvite=yes <- no space allowed |
| 16:46.04 | limb_q | hmm, i'll check |
| 16:46.05 | [TK]D-Fender | nat=yes <- for for THIS entry. |
| 16:46.17 | [TK]D-Fender | if you're behind NAT, please read THIS : |
| 16:46.19 | [TK]D-Fender | ~sipnat |
| 16:46.19 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:46.26 | [TK]D-Fender | decobbb, Yes, what about it? |
| 16:46.30 | shareenergy | anyone can help me to configure fax to email on a trixbox ? |
| 16:46.39 | shareenergy | i have a zaphfc |
| 16:46.58 | [TK]D-Fender | ~trixbox |
| 16:46.58 | jbot | methinks trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 16:47.09 | [TK]D-Fender | shareenergy, You are in the WRONG channel... please read the topic |
| 16:47.15 | limb_q | nat=yes i have |
| 16:47.40 | Voicemeup | hmm |
| 16:47.42 | Voicemeup | was .16 buggy ? |
| 16:47.43 | [TK]D-Fender | limb_q, you should NOT have "nat=yes" in your [xs4all] section. |
| 16:47.59 | limb_q | hmm |
| 16:48.06 | limb_q | but i'm behind router..? |
| 16:48.16 | [TK]D-Fender | Voicemeup, Naw... we just though it'd be nice to jump right to .24 for no reason ;) |
| 16:48.18 | *** join/#asterisk g1powermac (n=g1powerm@adsl-210-184-158.sdf.bellsouth.net) |
| 16:48.19 | shareenergy | how i define fax extension? |
| 16:48.22 | g1powermac | Hi All |
| 16:48.34 | limb_q | should i remove it of should i set it to no |
| 16:48.42 | [TK]D-Fender | limb_q, set to no |
| 16:48.50 | Voicemeup | Executing VoiceMailMain , -> then i get Playing 'vm-login' (language 'fr') but no sound on that... it comes from a main stanza inbound with a background greeting then you dial 3 digit for vmadmin |
| 16:49.02 | shareenergy | exten => _.,1,Goto(ext-fax,in_fax,1) |
| 16:49.02 | shareenergy | exten => _.,1,Noop(Entering zapincoming with DID = ${DID}) |
| 16:49.02 | shareenergy | exten => _.,1,Set(__FROM_DID=_.) |
| 16:49.02 | shareenergy | exten => _.,n,Gosub(app-blacklist-check,s,1) |
| 16:49.02 | shareenergy | exten => _.,n,Set(FAX_RX_EMAIL=fboleto@gmail.com) |
| 16:49.03 | shareenergy | exten => _.,n,Goto(ext-group,300,1) |
| 16:49.03 | g1powermac | does anyone know what would be the best VoIP service I could integrate into an asterisk pbx? |
| 16:49.04 | Voicemeup | the vm-password.gsm works.. and the file can be read |
| 16:49.05 | shareenergy | ups |
| 16:49.07 | [TK]D-Fender | shareenergy, Ok, that is clearly not relevant here. Go check out Trixbox's support channels & pages |
| 16:49.13 | limb_q | nothing |
| 16:49.15 | [TK]D-Fender | shareenergy, and do NOT spam in here |
| 16:49.27 | shareenergy | sorry |
| 16:49.29 | g1powermac | I was thinking of using BroadVoice, but wanted to see what you guys thought |
| 16:49.49 | shareenergy | is was a just stupid copy paste |
| 16:49.57 | limb_q | where should i start to cut this problem? |
| 16:50.00 | [TK]D-Fender | g1powermac, teliax comes better recommended around here. |
| 16:50.08 | limb_q | is there a logfile to see the handshake? |
| 16:50.11 | [TK]D-Fender | limb_q, pleaswe fully read the article I linked you. |
| 16:50.19 | limb_q | i'll check |
| 16:50.39 | g1powermac | [TK]D-Fender, k, thanks for the suggestion |
| 16:51.14 | [TK]D-Fender | g1powermac, Broadvoice's SIP proxies occasionally blink out. Have had mixed reviews from one customer of mine |
| 16:51.51 | g1powermac | yea, I heard about that |
| 16:52.53 | limb_q | i read it.. |
| 16:53.03 | g1powermac | [TK]D-Fender, hmm, what are the simultaneous calls bit with teliax? |
| 16:53.15 | limb_q | but my asterisk is in my own network behind my router to my provider |
| 16:53.20 | g1powermac | I can have upto so many calls using one number? |
| 16:53.22 | [TK]D-Fender | g1powermac, ? |
| 16:53.31 | limb_q | i have some sip credentials from my sip provider |
| 16:53.34 | g1powermac | https://www.teliax.com/newaccount/?r=1&cp=default |
| 16:53.50 | [TK]D-Fender | g1powermac, Not sure, go check them out for the particulars.. I have not used them personally, just heard opinions here from those I respect |
| 16:54.01 | g1powermac | [TK]D-Fender, ahh, ok |
| 16:54.05 | decobbb | tk: I was able to get green lights on my t1 card last night |
| 16:54.19 | decobbb | but the LEC said that they were unable to bring up the d channel |
| 16:54.20 | [TK]D-Fender | limb_q, Follow my guide for the rest of what you need for * to work behind NAT. |
| 16:54.38 | [TK]D-Fender | decobbb, pastebin your zaptel & zapata. |
| 16:54.43 | decobbb | ok |
| 16:54.51 | limb_q | i'm reading : http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:54.58 | [TK]D-Fender | limb_q, read the FIRST one. |
| 16:55.01 | [TK]D-Fender | ~sipnat |
| 16:55.01 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:55.09 | limb_q | my scenario is number 1 |
| 16:55.20 | limb_q | the first one is a different scenario |
| 16:55.39 | [TK]D-Fender | limb_q, Your * is behind NAT, thats for the first one covers. |
| 16:55.56 | [TK]D-Fender | ok, gotta go, back in a while. |
| 16:56.07 | [TK]D-Fender | decobbb, someone will be able to pick up your issue once PB'd |
| 16:56.36 | decobbb | http://pastebin.com/ma944ed8 |
| 16:56.48 | limb_q | but my sip clients are also behind nat |
| 16:56.54 | limb_q | only my sip provider not |
| 16:58.05 | decobbb | http://pastebin.com/m5555af27 |
| 16:59.58 | Voicemeup | oh well |
| 17:00.15 | limb_q | but is there a way that i can check if the connection went well? |
| 17:03.33 | limb_q | brb |
| 17:03.49 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 17:04.54 | *** join/#asterisk saftsack (n=oliver@p54A7E4B5.dip.t-dialin.net) |
| 17:05.56 | *** join/#asterisk michaelo (n=michaelo@adsl-068-159-111-129.sip.gsp.bellsouth.net) |
| 17:10.28 | *** join/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 17:10.41 | limb_q | back |
| 17:10.51 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 17:16.18 | limb_q | do i need to open some ports on my router to my *? |
| 17:19.46 | karleeto | limb_q: maybe 22 for ssh, and you may need to open some ports if you have trunks to other * boxes |
| 17:21.08 | decobbb | did anybody get a chance to look at my pastebin files for my zaptel.conf and zapata.conf |
| 17:21.14 | decobbb | http://pastebin.com/m5555af27 |
| 17:21.15 | decobbb | http://pastebin.com/ma944ed8 |
| 17:21.52 | decobbb | reguarding my d channels not coming up last night .. yet zttool said all was ok and i had green lights |
| 17:26.12 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 17:28.04 | limb_q | nope, only my sip connection to my provider |
| 17:30.59 | g1powermac | hmm, teliax looks quite good |
| 17:31.48 | rudholm | I'm very happy with teliax |
| 17:32.41 | rudholm | they're not the cheapest, but they're very reliable and I've never had any call quality issues |
| 17:33.01 | rudholm | I use their 2 cents per minute service for call completion |
| 17:33.08 | g1powermac | rudholm, well, they're the cheapest I've seen thus far for business solution I'm working on :-) |
| 17:33.16 | rudholm | ah |
| 17:33.22 | g1powermac | very, very cheap |
| 17:33.29 | rudholm | very cool then |
| 17:33.38 | rudholm | what's your need? |
| 17:34.13 | g1powermac | basically unlimited inbound and outbound calling including long distance, a few lines, and fax ability |
| 17:35.11 | rudholm | are you looking at the "Pay as you GO" or the "Corporate" product they offer? |
| 17:35.18 | g1powermac | the Corporate one |
| 17:35.35 | g1powermac | I love the unlimited incoming toll-free ability too |
| 17:35.39 | rudholm | the only limitation there is the maximum of four concurrent calls |
| 17:35.46 | rudholm | yeah, that's nice |
| 17:35.54 | rudholm | I don't use them for inbound |
| 17:36.07 | g1powermac | yea, but I could get a second line for only $9.99 a month |
| 17:36.29 | g1powermac | which I prolly will |
| 17:36.43 | rudholm | I think that 9.99 would get you an aditional phone number, I'm not sure it will get you four more simultaneous channels |
| 17:37.01 | rudholm | in fact, I don't think it will |
| 17:37.10 | g1powermac | hmm, will have to ask them about that |
| 17:37.12 | rudholm | but it's worth asking them |
| 17:37.13 | rudholm | yeah |
| 17:37.25 | rudholm | but you could get "Corporate" and a "Pay as you GO" |
| 17:37.37 | rudholm | since there's no monthly fee associated with Pay as you GO |
| 17:37.38 | *** join/#asterisk saftsack (n=oliver@p54A7D947.dip.t-dialin.net) |
| 17:37.50 | g1powermac | very true |
| 17:37.50 | rudholm | you could just send your overflow outbound calls out through your Pay as you GO plan. |
| 17:39.10 | *** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima) |
| 17:39.13 | g1powermac | yea, that definitely could be an option |
| 17:39.16 | rudholm | I really like PAYG, since I'm not a heavy phone user. they end up charging me credit card a 10$ refill every month or two |
| 17:39.25 | rudholm | and they're pretty good about letting you set the outbound CID |
| 17:39.53 | g1powermac | yea, that plan is ideal if you don't use the phones much |
| 17:40.05 | g1powermac | though in this case, the phones are used quite a bit :-) |
| 17:40.08 | rudholm | you can set it to any NXXNXXXXXX you want, but you can't set it to something "weird" |
| 17:40.23 | rudholm | yeah, for a business, it's different. |
| 17:40.49 | g1powermac | thats cool |
| 17:40.53 | rudholm | but if you stay under 5 concurrent calls most of the time, it probably makes sense to get one of each account tyep |
| 17:42.01 | g1powermac | yea |
| 17:42.15 | g1powermac | though I think really the four concurrent calls will be plenty |
| 17:42.21 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 17:42.23 | rudholm | since getting two Corporate accounts would be 90$/month and would still limit you to 8 calls. that extra 45$ would buy 2250 minutes from the pay as you go account. |
| 17:42.41 | g1powermac | our business only uses currently two analog lines |
| 17:42.56 | rudholm | oh, then yeah, Corporate would be plenty |
| 17:43.21 | g1powermac | but it leaves some nice expansion room for later heavier use |
| 17:43.25 | rudholm | yeah |
| 17:43.36 | rudholm | are you close to one of their points of presence? |
| 17:43.41 | g1powermac | not sure |
| 17:43.46 | g1powermac | do they have a map? |
| 17:44.11 | rudholm | not sure |
| 17:44.22 | rudholm | I'm in L.A. so I use voip-ca1.teliax.com |
| 17:44.25 | g1powermac | I know they can port the current numbers we have |
| 17:44.37 | rudholm | voip-co2.teliax.com is in colorado |
| 17:44.39 | g1powermac | which is a very good thing (tm) :-) |
| 17:44.43 | rudholm | I think they have east coast too |
| 17:44.44 | rudholm | yes |
| 17:45.02 | rudholm | also, they seem stable/ethical enough that I might actually trust them with numbers I value. |
| 17:45.48 | rudholm | I have a BroadVoice account too, but I'd *never* port numbers to them. |
| 17:45.57 | rudholm | they're completely shady |
| 17:46.08 | g1powermac | hmm, good thing I didn't go that route |
| 17:46.34 | g1powermac | I was thinking of using the vonage business plus plan thingie |
| 17:46.36 | g1powermac | http://www.vonage-business-plus.com/ |
| 17:46.42 | rudholm | most ITSPs are about as stable as a dot-com startup in 2000 |
| 17:47.07 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 17:47.16 | g1powermac | but was a bit concerned with vonage and the mega law suit they're under |
| 17:47.28 | *** join/#asterisk ManxPower (n=manxpowe@015-850-242.area5.spcsdns.net) |
| 17:47.57 | rudholm | wow, 150$ ? |
| 17:48.24 | g1powermac | yea, thats why I thought teliax was so cheap :-) |
| 17:48.48 | rudholm | I was a vonage customer from 2003 to 2007 and can tell you, they have no advantage over Teliax, and some disadvantages. |
| 17:49.33 | g1powermac | yea, we currently have vonage at home, and it isn't the best, but does serve the purpose |
| 17:49.52 | rudholm | yeah. I cancelled because they wouldn't open the SIP credentials |
| 17:50.03 | *** part/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:50.32 | rudholm | but they're really inflexible |
| 17:50.33 | rudholm | in general |
| 17:50.41 | g1powermac | yea |
| 17:50.47 | rudholm | and you can't set outbound CID of course |
| 17:51.07 | rudholm | so if you forward calls through your Asterisk, you'll lose the original CID |
| 17:51.20 | rudholm | ATA into an FXO? ew. |
| 17:51.44 | g1powermac | yea, it was just for testing |
| 17:51.58 | g1powermac | I was originally going to hook up the analog lines we got here to it |
| 17:52.11 | g1powermac | but then we found out we can get decent cable lines run to our store and warehouse/office and could actually go with a full VoIP solution |
| 17:52.29 | g1powermac | and finally stop paying the monster that is At&t/Bellsouth |
| 17:52.33 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 17:52.58 | rudholm | ah, here's their east coast facility: voip-ny1.teliax.com |
| 17:53.07 | rudholm | see what your ping times are to all three |
| 17:53.15 | g1powermac | so all the work done for the asterisk box was well, for moot |
| 17:53.15 | rudholm | the one in Colorado is the main facility |
| 17:53.38 | g1powermac | but at least we don't have to deal with at&t anymore :-) |
| 17:53.52 | g1powermac | ahh, cool |
| 17:53.56 | rudholm | but you're still going to use Asterisk, right? |
| 17:54.01 | g1powermac | yup |
| 17:54.15 | g1powermac | but will be starting from scratch |
| 17:55.24 | rudholm | it should be pretty similar, you're just using an IAX peer instead of FXO ports |
| 17:55.33 | rudholm | most of your extensions.conf will be the same |
| 17:56.01 | g1powermac | well, we're also converting all the phones to SIP phones :-) |
| 17:56.11 | g1powermac | yea, the old system was all analog phones too |
| 17:56.27 | rudholm | ah |
| 17:56.53 | rudholm | my employer is planning to move to Asterisk as well I recently found |
| 17:57.23 | rudholm | it's going to be a pretty big project |
| 17:57.23 | *** join/#asterisk CCFL_Man2 (i=7599fc12@pool-71-241-87-104.scr.east.verizon.net) |
| 17:57.29 | g1powermac | and because the analog phone lines here were so bad, asterisk's echo cancelers couldn't handle it |
| 17:58.11 | g1powermac | so for the short time the old system was up, I was pulling my hair out of my head trying to fix the echo problems |
| 17:58.17 | CCFL_Man2 | g1powermac: asterisk on a g1 power mac? |
| 17:58.30 | ManxPower | g1powermac: So how are you going to solve the echo problems? |
| 17:58.37 | g1powermac | CCFL_Man2, nah, though that would be interesting |
| 17:58.51 | g1powermac | ManxPower, no more analog at all :-) |
| 17:58.52 | CCFL_Man2 | yeah |
| 17:59.10 | g1powermac | going completely digital here |
| 17:59.13 | g1powermac | bbiab |
| 17:59.23 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 18:00.00 | CCFL_Man2 | i just got a cisco mc3810 |
| 18:00.04 | ManxPower | g1powermac: that will not work. |
| 18:00.46 | ManxPower | Most echo is caused by the FAR end analog line. So unless you are planning on never calling an analog phone, you need at actual, real, working echo canceler. |
| 18:01.25 | CCFL_Man2 | now, can i connect a multiflex T1 port to a channel bank? |
| 18:01.57 | ManxPower | CCFL_Man2: Multiflex is a marketing term. What is the technical term |
| 18:02.30 | *** join/#asterisk limb_q (n=root@limbique.xs4all.nl) |
| 18:02.33 | limb_q | back |
| 18:02.52 | CCFL_Man2 | ManxPower: not sure, there are two interface cards for the mc3810, the MFT-T1 card and the DVM-T1 card, the latter is used for connecting to pbx's and channel banks |
| 18:03.14 | ManxPower | CCFL_Man2: find out. |
| 18:03.46 | ManxPower | Well if it is going into generic channel banks then a good term is Channelized Voice T-1 |
| 18:03.49 | limb_q | anyone has some good experience with sangoma? |
| 18:04.10 | ManxPower | limb_q: The work, they can be a bitch to set up |
| 18:04.20 | limb_q | hmm |
| 18:04.32 | CCFL_Man2 | the MFT-T1 is multiflex trunking and BRI |
| 18:04.36 | limb_q | we have some problems they drops d-channel on pri |
| 18:04.55 | limb_q | or is it bri |
| 18:05.09 | limb_q | 30 channel isdn card |
| 18:05.41 | limb_q | gonna eat now.. |
| 18:05.46 | limb_q | brb l8er |
| 18:05.58 | limb_q | part |
| 18:06.03 | *** part/#asterisk limb_q (n=root@limbique.xs4all.nl) |
| 18:07.03 | CCFL_Man2 | the mc3810 i'm getting has a MFT-T1 card |
| 18:07.27 | ManxPower | CCFL_Man2: Might you mean PRI? |
| 18:07.48 | CCFL_Man2 | i think so |
| 18:10.06 | CCFL_Man2 | Multiflex Trunk Modules with Optional BRI |
| 18:10.16 | CCFL_Man2 | <PROTECTED> |
| 18:10.16 | CCFL_Man2 | <PROTECTED> |
| 18:10.18 | CCFL_Man2 | MC3810-MFT-T1 |
| 18:10.23 | crimethinker | men, why must you lie? |
| 18:10.53 | elixer | women, why must you ask dumb questions? |
| 18:10.56 | elixer | heh |
| 18:11.44 | *** join/#asterisk sysreq (n=sysreq@bas9-montrealak-1096749256.dsl.bell.ca) |
| 18:15.49 | *** join/#asterisk saftsack (n=saftsack@pD9E0530D.dip.t-dialin.net) |
| 18:16.34 | ManxPower | CCFL_Man2: then the docs are wrong. BRIs hace TWO voice channels and ONE signalling channel. |
| 18:20.42 | *** join/#asterisk Tmob (n=total@c-24-6-119-95.hsd1.ca.comcast.net) |
| 18:21.40 | *** part/#asterisk Tmob (n=total@c-24-6-119-95.hsd1.ca.comcast.net) |
| 18:22.38 | CCFL_Man2 | ManxPower: i must be wrong |
| 18:22.55 | ManxPower | CCFL_Man2: someone is wrong. |
| 18:23.02 | *** join/#asterisk hohum_ (n=dcorbe@dhcp64-134-231-200.shs.nyc.wayport.net) |
| 18:23.06 | ManxPower | CCFL_Man2: Are you in the USA or Canada? |
| 18:23.07 | CCFL_Man2 | in any case i don't think that card will work for me |
| 18:23.47 | CCFL_Man2 | i'm going to be getting voice from the mc3810, not putting voice into it |
| 18:24.08 | *** join/#asterisk Jmarcu (n=total@c-24-6-119-95.hsd1.ca.comcast.net) |
| 18:24.50 | CCFL_Man2 | US |
| 18:24.52 | g1powermac | ManxPower, yea, I know that, however, the echo canceler was having alot of problems with the bad analog lines we have, which won't be used when we switch to VoIP |
| 18:24.53 | Jmarcu | hi.. anyone here can recommend a good voip phone service (like vonage?) ... i'm planning to send one of those units to my parents overseas so they can call US for cheap.. |
| 18:25.55 | ManxPower | g1powermac: Using digital lines (T-1, PRI, VoIP) will eliminate some sources of echo. |
| 18:25.55 | rudholm | g1powermac: yeah, I can tell you from experience that analog lines on your end can at least contribute to echo problems |
| 18:25.55 | ManxPower | I would have recommended you go with a PRI, as they are more reliable for most people. |
| 18:26.04 | rudholm | I think PRI is outside his budget |
| 18:26.23 | g1powermac | yea, so is a T1 |
| 18:26.28 | CCFL_Man2 | Jmarcu: vonage is not good |
| 18:27.28 | rudholm | I have one FXO port and used to have it on a TDM400 card. echo was horrible --basically unusable. I switched to a TDM800 (with the same FXO module) and now there's no echo |
| 18:27.32 | Jmarcu | CCFL_Man2, ic.. anything you recommend i checkout? |
| 18:27.54 | CCFL_Man2 | Jmarcu: definately quantumvoice, i personally use it |
| 18:28.29 | ManxPower | g1powermac: you'll find your budget increased when your call quality sucks everytime someone sends a large attachement on an e-mail |
| 18:28.59 | g1powermac | ManxPower, thats where QoS comes in :-) |
| 18:29.16 | *** join/#asterisk crimethinker (n=ircuser@legacy.diamond.org) |
| 18:29.30 | g1powermac | ManxPower, I got openwrt running routers that have a vpn link setup between the two locations we have |
| 18:29.31 | ManxPower | g1powermac: you can't QoS data that goes across networks you do not manage. |
| 18:29.50 | Jmarcu | CCFL_Man2, oh.. ok lemme search |
| 18:29.54 | CCFL_Man2 | g1powermac: why not bitch and barter to the telco to fix your analog lines? |
| 18:29.59 | g1powermac | that is true, but I could do it on our internal network |
| 18:30.15 | ManxPower | You can improve latency and jitter by managing TCP connections, but other than that it won't work. |
| 18:30.16 | CCFL_Man2 | Jmarcu: quantumvoice.com |
| 18:30.19 | ManxPower | g1powermac: how are your calls going to the outside world? |
| 18:30.28 | g1powermac | CCFL_Man2, we're still trying to get them to give us the right bill amounts |
| 18:30.44 | g1powermac | CCFL_Man2, its been a two year fight with that |
| 18:30.54 | CCFL_Man2 | g1powermac: hah, typical of Ma'Bell |
| 18:31.06 | g1powermac | ManxPower, I'm looking at Teliax |
| 18:31.30 | ManxPower | g1powermac: They are a good provider, but your calls are going over the internet so you can't QoS them. |
| 18:31.50 | ManxPower | If your users don't hate you every time there is a blip in the audio then you should be OK. |
| 18:31.53 | g1powermac | CCFL_Man2, yea, we filed complaints with the local utility commission here as well, and even threatened law suits |
| 18:32.52 | CCFL_Man2 | most of the time voip over the internet works fairly well, only incomming qos you can do is get a dedicated line for it |
| 18:33.05 | CCFL_Man2 | g1powermac: typical |
| 18:33.05 | Jmarcu | CCFL_Man2, but their unlimited plan is 29.99.. more expensive than others it seems right? |
| 18:33.30 | g1powermac | ManxPower, a blip is a bit better than what we have now |
| 18:33.44 | CCFL_Man2 | Jmarcu: yeah, i went with the $19.99 plan |
| 18:33.46 | g1powermac | ManxPower, the analog systems are so bad, we've had echos without any pbx attached |
| 18:34.11 | g1powermac | and the dsl we get from them is utter crap |
| 18:34.14 | ManxPower | g1powermac: I understand. |
| 18:34.32 | Jmarcu | CCFL_Man2, hmm.. yea thats not too bad actally.. 1000 minutes is pretty lot for my parents anyway |
| 18:34.34 | ManxPower | by going all VoIP you have eliminated echo on your end. By using an ITSP you should handle echo on the far side |
| 18:34.54 | g1powermac | with an average line attenuation of about 56db, its really bad |
| 18:34.55 | ManxPower | But I say that sending calls over the internet is not the way to have reliable phone service. |
| 18:35.23 | g1powermac | yea, that is a risk I told the others about |
| 18:36.19 | g1powermac | same really goes for the internet connection |
| 18:36.37 | decobbb | does anybody have time to look at my pastebin files for my zaptel.conf and zapata.conf |
| 18:36.41 | g1powermac | since it isn't a t1, there isn't guarantees for uptime |
| 18:36.44 | decobbb | reguarding my d channels not coming up last night .. yet zttool said all was ok and i had green lights |
| 18:36.50 | decobbb | http://pastebin.com/ma944ed8 |
| 18:36.52 | decobbb | http://pastebin.com/m5555af27 |
| 18:38.42 | ManxPower | decobbb: you have more than one channel 1 defined. |
| 18:38.56 | ManxPower | Asterisk channels start at 1 and up. they never repeat |
| 18:39.35 | decobbb | in witch file? |
| 18:40.37 | ManxPower | <PROTECTED> |
| 18:40.46 | ManxPower | ztcfg -vvv should have generated errors with that file. |
| 18:41.27 | decobbb | ok |
| 18:41.33 | decobbb | fixed that |
| 18:41.49 | decobbb | do you think that would keep the d channel from coming up ? |
| 18:42.21 | decobbb | my apologies for my ignorance |
| 18:42.29 | g1powermac | rudholm, http://www.teliax.com/forum/viewtopic.php?t=871 |
| 18:42.41 | decobbb | im learning telephony coming from the newtork engineer world |
| 18:42.46 | g1powermac | rudholm, seems the extra lines do mean more simultaneous calls :-) |
| 18:42.53 | ManxPower | decobbb: "pri debug span X" where span X is the one going to your telco. |
| 18:43.13 | decobbb | wouldnt they both go out to the telco ? |
| 18:43.16 | ManxPower | If you don't get any data, then call the telco and say "I have no traffic on the D-channel, fix it" |
| 18:43.20 | decobbb | i have 2 pri's? |
| 18:43.23 | *** join/#asterisk michaelo (n=michaelo@adsl-068-159-111-129.sip.gsp.bellsouth.net) |
| 18:44.03 | decobbb | ok |
| 18:44.27 | decobbb | do i need to make a seperate group ffor each pri ? |
| 18:44.38 | ManxPower | decobbb: http://pastebin.com/m5555af27 It shows 2 d channels |
| 18:44.39 | decobbb | or is it ok to group them into 1 |
| 18:45.00 | ManxPower | 2 D-channels in my book means 2 PRIs |
| 18:45.07 | decobbb | http://pastebin.com/ma944ed8 |
| 18:45.08 | decobbb | on this one |
| 18:45.14 | ManxPower | Sicne you have them configured as PRI NET you are are acting as the telco |
| 18:45.16 | decobbb | do i need to define 2 groups |
| 18:45.35 | ManxPower | that one is WRONG WRONG WRONG |
| 18:45.38 | decobbb | so they should be set to pr_cpe |
| 18:45.45 | decobbb | pri_cpe |
| 18:45.48 | ManxPower | decobbb: group= is for dialing OUT and does not group channels for signalling |
| 18:46.11 | ManxPower | you STILL have channel 1 defined twice in the file |
| 18:46.16 | decobbb | ok |
| 18:46.25 | decobbb | i switched that one to channel 49 |
| 18:46.49 | ManxPower | Dial(Zap/g1/5551515) will dial out using the channels in group=1, starting with the lowest available channel |
| 18:47.08 | ManxPower | decobbb: what exactly DO you have? |
| 18:47.22 | decobbb | i have 2 pri |
| 18:47.29 | decobbb | i have a 2 port pri card |
| 18:47.36 | decobbb | <PROTECTED> |
| 18:48.03 | ManxPower | and what lines do you have? |
| 18:48.18 | decobbb | 2 23 channel pri cards |
| 18:48.25 | decobbb | sorry |
| 18:48.26 | ManxPower | so you have 2 pris? |
| 18:48.28 | decobbb | yes |
| 18:48.34 | wothinn | Does it take a certain load before musiconhold requires a timing source? I'm running on OpenBSD, so none available, but MOH seems to be working flawlessly for me between two local SIP phones. |
| 18:48.36 | ManxPower | and those two PRIs go to the telco? |
| 18:48.39 | decobbb | yes |
| 18:48.57 | wothinn | I'm just curous if this'll all fall apart if I increase the load some. |
| 18:48.58 | ManxPower | so you have 2 PRIs going to the telco and 24 analog lines going where? |
| 18:49.10 | decobbb | to internal phones |
| 18:50.05 | ManxPower | forget about the analog lines right now. |
| 18:50.12 | decobbb | ok |
| 18:50.16 | ManxPower | try this http://pastebin.com/m70d4b7fb |
| 18:50.50 | ManxPower | and this http://pastebin.com/m419180b7 |
| 18:53.20 | wothinn | Oh, scratch that... it's starting to get choppy after a few minutes on hold. |
| 18:53.46 | ManxPower | decobbb: this is more correct http://pastebin.com/m310dca31 |
| 18:53.55 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 18:53.58 | decobbb | ok |
| 18:55.19 | ManxPower | using the files I provided, what is the output of ztcfg -vvv ? Put the info on pastebin |
| 18:57.15 | decobbb | http://pastebin.com/m43724598 |
| 18:57.40 | decobbb | is the output |
| 18:59.40 | g1powermac | hmm, does anyone know if teliax offers a way to send the voip data to them encrypted? |
| 19:00.18 | ManxPower | decobbb: Good! Now does asterisk start? |
| 19:00.24 | decobbb | yep |
| 19:00.32 | ManxPower | g1powermac: I don't think they do. Most ITSPs don't. |
| 19:00.45 | ManxPower | decobbb: does "pri debug span 1" show data |
| 19:00.48 | g1powermac | k |
| 19:01.10 | *** join/#asterisk limb_q (n=root@limbique.xs4all.nl) |
| 19:01.12 | limb_q | howdy |
| 19:01.57 | decobbb | i cannot so that for another couple of hours |
| 19:01.57 | decobbb | "after hours" |
| 19:01.57 | decobbb | if it does |
| 19:01.57 | decobbb | what should i look for anytype of data? |
| 19:02.50 | ManxPower | decobbb: you should see all sort of technical crap. |
| 19:02.55 | decobbb | ok |
| 19:02.59 | ManxPower | that would indicated it is prolly working. |
| 19:03.17 | decobbb | and if i see nothing tell the telco i dont see anything |
| 19:03.20 | ManxPower | also do a "pro no debug span 1" and a "pri debug span 2" to make sure you have data there too. |
| 19:03.22 | ManxPower | correct. |
| 19:03.29 | *** join/#asterisk humbertopt (n=ss@10001344683.0000074353.acesso.oni.pt) |
| 19:03.40 | *** join/#asterisk nahirean (n=nahirean@unaffiliated/nahirean) |
| 19:03.45 | decobbb | cool |
| 19:03.51 | decobbb | thanx so much |
| 19:03.55 | humbertopt | Hi 2 all! |
| 19:05.44 | humbertopt | Can someone help with a misdn problem? |
| 19:09.38 | *** join/#asterisk Weezey (n=ohno@206.210.109.232) |
| 19:11.58 | rudholm | is quamtumvoice.com good? |
| 19:12.05 | humbertopt | I'm running asterisk with a Beronet card since some months. Everything was ok until this week: Most of my dialout calls are ended with code 31... |
| 19:13.07 | humbertopt | If I use a different provider (dialing a prefix) it works ok... |
| 19:13.27 | ManxPower | humbertopt: make sure the caller id you send does not start with 0, 00 or 1 |
| 19:13.42 | humbertopt | Ok |
| 19:13.48 | humbertopt | I will check that |
| 19:14.20 | humbertopt | But I think my telco allways change my caller id somehow |
| 19:16.05 | humbertopt | With my old pbx everything works, so I can't really complain with my telco about this |
| 19:16.43 | ManxPower | humbertopt: and it worked with Asterisk as well until this week. |
| 19:16.44 | decobbb | question |
| 19:17.01 | decobbb | does *97 work by default to check Voicemail |
| 19:17.16 | ManxPower | decobbb: not unless you set it up that way |
| 19:17.26 | decobbb | ok |
| 19:17.31 | decobbb | i was getting congestion |
| 19:17.35 | ManxPower | Asterisk really isn't a PBX. It is more of a PBX Toolkit that lets you build a PBX |
| 19:17.38 | decobbb | just wanted to make sure |
| 19:18.09 | *** join/#asterisk karleeto_lap (n=karl@techwifi.franklincomputer.net) |
| 19:18.23 | *** part/#asterisk karleeto_lap (n=karl@techwifi.franklincomputer.net) |
| 19:20.16 | *** join/#asterisk shareenergy (n=shareene@195-23-137-26.net.novis.pt) |
| 19:20.27 | shareenergy | is beronet.com down? |
| 19:20.41 | shareenergy | is there any mirror where i can take the misdn? |
| 19:20.55 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 19:21.27 | decobbb | how do it set it to pass on the extension calling it so you just have to enter the password? |
| 19:22.06 | decobbb | for *97 voicemail |
| 19:22.55 | decobbb | right now i have it pointing to VoicemailMain |
| 19:24.32 | [TK]D-Fender | decobbb, you can pass the box # as a parameter. As to "knowing" who's calling for it... think CALLER-ID |
| 19:25.09 | decobbb | so if i have my caller id set to names |
| 19:25.16 | decobbb | it will not pass the extension number |
| 19:25.22 | decobbb | correct ? |
| 19:25.29 | [TK]D-Fender | decobbb, caller Id hold name AND number... |
| 19:25.47 | shareenergy | [TK]D-Fender do you know where i can get the install-misdn-mqueue.tar.gz ? |
| 19:25.52 | [TK]D-Fender | shareenergy, nope. |
| 19:26.13 | decobbb | caller_id=decobbb 3872 |
| 19:26.18 | decobbb | would that work ? |
| 19:26.44 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) [NETSPLIT VICTIM] |
| 19:29.43 | [TK]D-Fender | decobbb, close |
| 19:33.17 | decobbb | tk: what am i missing ? |
| 19:34.57 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) [NETSPLIT VICTIM] |
| 19:35.05 | [TK]D-Fender | decobbb, formatting is off in a few places. http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf |
| 19:41.24 | humbertopt | No luck |
| 19:41.42 | humbertopt | everythink seams ok with caller id |
| 19:41.50 | humbertopt | everything seams ok with caller id |
| 19:41.57 | ManxPower | decobbb: callerid would be callerid=Name <number> |
| 19:43.17 | humbertopt | It seams my telco is taking a lot of time to make calls (more than 15 sec.) |
| 19:43.43 | humbertopt | maybe misdn dosen't like this |
| 19:43.46 | ManxPower | if you had callerid=decobbb 3872 then your callerid NAME would be "decobbb 3872" and your callerid number would be empty |
| 19:46.34 | humbertopt | Where's the best place to put my logs and get some help? |
| 19:52.48 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 19:52.52 | humbertopt | Where's the best place to put my logs and get some help? |
| 19:55.47 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) [NETSPLIT VICTIM] |
| 20:05.59 | ManxPower | ~pastebin |
| 20:06.00 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 20:06.08 | crimethinker | make it stop |
| 20:07.14 | CCFL_Man2 | make teh pain stop? |
| 20:12.49 | humbertopt | http://pastebin.ca/654807 |
| 20:13.15 | humbertopt | Can anyone help? Thanks! |
| 20:22.11 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 20:24.24 | g1powermac | I got this one for the wifi network I have setup: http://www.voipsupply.com/product_info.php?products_id=802{73}380{9}25{15}42{17}49 |
| 20:26.35 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:34.46 | [TK]D-Fender | g1powermac, gotten a lot of really bad reviews |
| 20:36.00 | g1powermac | [TK]D-Fender, really? like what? |
| 20:36.49 | [TK]D-Fender | g1powermac, crappy range, 802.11b (low speed) drags down your router, slow ping response (on of my clients running them had constant time-out on qualify), etc. |
| 20:36.59 | [TK]D-Fender | low battery life was one major point |
| 20:38.02 | g1powermac | well, the range won't be too much of a problem, I have upgraded the wifi here to the point that my horrifically bad powerbook in the terms of wifi reception does quite well |
| 20:38.30 | g1powermac | the phones does to 802.11g and will only be able to connect to my network via that |
| 20:38.40 | g1powermac | s/to/do |
| 20:39.12 | g1powermac | and for battery life, that could be an issue |
| 20:39.55 | [TK]D-Fender | g1powermac, Guess you'll see soon enough... |
| 20:40.19 | g1powermac | yea |
| 20:41.42 | g1powermac | I only bought one, so if it doesn't work well, I could get others |
| 20:42.42 | g1powermac | though you should see the wifi antennas I got for my routers: http://www.fab-corp.com/product.php?productid=3069&cat=250&page=1 :-) |
| 20:42.52 | g1powermac | they're about 2.5 times as big as the standard ones |
| 20:47.54 | [TK]D-Fender | ~wifisip |
| 20:47.55 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 20:47.57 | [TK]D-Fender | ^^^^^^^^^ |
| 20:48.37 | g1powermac | hmm |
| 20:48.49 | wishes | man i dont think ive seen anyone say that a particular phone doesnt suck yet other than really expensive ones :) |
| 20:49.06 | wishes | 'grandstream sucks' 'soft phones suck' 'wifi phones suck' |
| 20:49.17 | *** join/#asterisk saftsack (n=saftsack@pD9E0530D.dip.t-dialin.net) |
| 20:49.32 | [TK]D-Fender | wishes, So far Polycom & cisco are safe :) |
| 20:49.42 | [TK]D-Fender | wishes, And hardly "expensive" |
| 20:49.50 | [TK]D-Fender | Polycom anyways.... |
| 20:50.52 | g1powermac | polycom, ehh? |
| 20:50.57 | [TK]D-Fender | yup |
| 20:51.14 | [TK]D-Fender | g1powermac, check out www.telephonydepot.com |
| 20:51.23 | [TK]D-Fender | Voipsupply = overpriced hassle |
| 20:52.06 | g1powermac | hmm, telephony depot doesn't have any polycom wifi sip phones |
| 20:52.15 | [TK]D-Fender | g1powermac, they don't MAKE any. |
| 20:52.22 | g1powermac | oh |
| 20:52.30 | [TK]D-Fender | g1powermac, Well actually.. they bought out a DECT phone maker, forgot the name |
| 20:52.44 | [TK]D-Fender | For wireless you're better off with an ATA + cordless phone. |
| 20:55.46 | hmmhesays | I smash right through your stop light |
| 20:57.57 | [TK]D-Fender | hmmhesays, http://www.youtube.com/watch?v=tiQVtvj323U |
| 21:00.54 | hmmhesays | LOL nice |
| 21:01.17 | *** join/#asterisk Tili (n=tili@153.Red-80-38-134.staticIP.rima-tde.net) |
| 21:02.46 | [TK]D-Fender | "I may not be able to turn right on a red light... but tabarnac I can go right throught it! |
| 21:02.59 | *** join/#asterisk pagec (n=pagec@cpe-74-73-191-68.nyc.res.rr.com) |
| 21:05.57 | pagec | with extensions.conf and globals asterisk would evaluted nested globals immediately, with ael it instead returns the string. i.e. in ael OUTERGLOBAL=${GLOBAL(innerglobal)}; and NoOp($GLOBAL(OUTERGLOBAL)}); yields ${GLOBAL(innerglobal)} and not the value of innerglobal. does anyone know how to make it so that the value of innerglobal is the result intead when an arbitrary about of nesting is used |
| 21:07.08 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:09.03 | decobbb | hey guys |
| 21:09.20 | decobbb | when i type the command pri debug span 1 |
| 21:09.32 | decobbb | it returns no pri running on span 1 |
| 21:09.47 | decobbb | any ideas? |
| 21:10.28 | decobbb | exit |
| 21:10.32 | decobbb | whoops |
| 21:11.11 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 21:12.34 | decobbb | zttool is showing all is ok |
| 21:13.04 | decobbb | should i call the lec? |
| 21:13.34 | hmmhesays | I'm downloading the new driver game |
| 21:14.38 | [TK]D-Fender | decobbb, pastebin your zaptel &zapata again |
| 21:15.14 | hmmhesays | ahh my speak to female 1/4 adapters came in the mail yesterday |
| 21:15.15 | hmmhesays | ROCK |
| 21:15.16 | Weezey | I'm getting no RTP out from my * machine when I call into a menu, I only see the RTP from the phone calling it. |
| 21:16.09 | hmmhesays | sounds like a networking issue |
| 21:17.09 | decobbb | http://pastebin.com/m13faef2f |
| 21:17.12 | decobbb | zapata.conf |
| 21:17.56 | Weezey | hmmhesays: I just realized that it worked yesterday after my upgrade until I loaded up the sangoma wanrouter shit. |
| 21:18.00 | decobbb | http://pastebin.com/m33282d06 |
| 21:18.03 | decobbb | zaptel.conf |
| 21:21.21 | hmmhesays | in 2 hours I will have drive 4 on this pc |
| 21:21.23 | hmmhesays | I'm pumped |
| 21:24.22 | decobbb | any ideas? |
| 21:25.56 | [TK]D-Fender | decobbb, BOTH |
| 21:26.18 | [TK]D-Fender | decobbb, And you should have your first designated as a PRIMARY timing source, and the other as secondary. |
| 21:26.21 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 21:26.34 | [TK]D-Fender | decobbb, http://pastebin.com/m1402fd28 |
| 21:27.14 | [TK]D-Fender | decobbb, lack of clock sync might kill your DChan |
| 21:27.34 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-116-35.red.bezeqint.net) |
| 21:27.43 | decobbb | ok |
| 21:27.52 | decobbb | so i put that in my zaptel.conf |
| 21:27.59 | decobbb | should i reload my zaptel service? |
| 21:29.33 | *** join/#asterisk Slingky (n=Slingky@modemcable199.182-200-24.mc.videotron.ca) |
| 21:29.45 | Slingky | hi guys, does somebody can help me with disa ? |
| 21:29.52 | decobbb | still returning no pri running on span 1 |
| 21:30.02 | decobbb | when i type in the debug command |
| 21:30.03 | [TK]D-Fender | decobbb, completely stop *, redo "ztcfg -vvvv" and restart * |
| 21:30.14 | decobbb | ok |
| 21:33.42 | wishes | [TK]D-Fender: heh thanks :D |
| 21:33.55 | wishes | cisco is out, ill look at the other one .. |
| 21:34.15 | wishes | i think we're gonna go mostly softphone except for customer service/account managers who use it all th etime :) |
| 21:35.54 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:39.03 | hmmhesays | softphones can be ok |
| 21:39.09 | *** join/#asterisk jpfjp (n=jpfjpf@193.4.117.141) |
| 21:39.18 | jpfjp | is this a newbie channel? |
| 21:39.48 | hmmhesays | have you read the book? |
| 21:40.46 | jpfjp | what book |
| 21:40.56 | hmmhesays | the asterisk book |
| 21:43.11 | jpfjp | nope |
| 21:44.05 | jpfjp | it seems that asterisk 1.2 is in security maintenance mode |
| 21:44.18 | jpfjp | when asterisk 1.4 isn't yet as stable as 1.2 |
| 21:45.48 | decobbb | same message no Pri running on span 1 |
| 21:47.07 | fujin | Haven't had any difficulty going to 1.4 from 1.2 myself |
| 21:47.23 | fujin | Made the change last friday, system has been stable since |
| 21:47.40 | fujin | I was running 1.2 from Ubuntu, so I feel a bit safer having built it myself. |
| 21:47.57 | fujin | Ported my dialplan to AEL - pure sex. |
| 21:50.11 | *** join/#asterisk bosman (i=bosman@prozac.bsdzine.org) |
| 21:51.04 | wothinn | Does anyone know why my Polycom IP500's display would be flashing after I lost power and it came back? All seems to be working, just the display is flashing. |
| 21:52.36 | fujin | tried power cycling it? |
| 21:52.41 | wothinn | Yep. |
| 21:53.00 | mvanbaak | tried hitting it with a fireaxe ? |
| 21:53.29 | wothinn | I would, but I like it too much. :) |
| 21:53.48 | decobbb | hey guys what file do you set the dnis in ? |
| 21:56.15 | wothinn | Aaah... release notes for the sip.ld... 7204: Added flashing time/date until successful SNTP response |
| 21:56.27 | wothinn | So my NTP server is pooched. That's easy. :) |
| 21:56.55 | fujin | oh, nice ;) |
| 21:56.59 | fujin | openntpd! |
| 21:57.06 | fujin | do your phones not set their time based on SIP? |
| 21:57.18 | fujin | my ones do sntp and fallback to SIP |
| 21:57.51 | wothinn | fujin: I'm using OpenNTPD (on OpenBSD)... It appears this phone falls back to SIP too since it knows the time... it's just flashing. |
| 21:57.58 | fujin | ah |
| 21:58.00 | fujin | odd, that |
| 21:58.13 | fujin | I run openntpd on everything too, it appears to be much better than the netkit-ntpd |
| 21:58.19 | wothinn | Looks like my NTPD is just waiting to sync better before it'll give an authoritative response. It's all good. |
| 21:59.01 | fujin | best thing I ever setup: centralised syslog via syslog-ng from my phones |
| 21:59.08 | fujin | makes debugging alot of stuff muche asier |
| 22:03.05 | hmmhesays | I hate web dev |
| 22:04.07 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 22:05.17 | mvanbaak | hmmhesays: me too. too bad it's my job ;) |
| 22:05.27 | hmmhesays | mvanbaak: its mine right now too |
| 22:05.38 | hmmhesays | writing a user interface for this phone system we're doing |
| 22:05.44 | hmmhesays | all I have to say is UGH |
| 22:05.58 | hmmhesays | javascript/ajax |
| 22:06.17 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:07.08 | mvanbaak | use jquery :) |
| 22:08.28 | hmmhesays | what is jquery? |
| 22:09.14 | mvanbaak | a javascript framework to make stuff like ajax easier |
| 22:09.15 | hmmhesays | nm I'll google |
| 22:13.41 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:17.43 | *** join/#asterisk rrittenhouse (n=tad@unaffiliated/rrittenhouse) |
| 22:19.53 | *** join/#asterisk asdx (n=diego@adsl-152-30.click.com.py) |
| 22:20.23 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 22:21.10 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 22:21.10 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 22:21.43 | *** join/#asterisk QbY (n=thatkidk@208.36.224.228.ptr.us.xo.net) |
| 22:22.17 | rrittenhouse | I'm trying to set up an asterisk machine for my work (Im learning about it at home before I take it to work) - I'm wondering is there a (cheaper) way to get a SIP VOIP service for testing asterisk instead of buying the hardware to hook it up to my phone system at home? |
| 22:22.24 | QbY | Anyone have a suggestion for speech recognition in the dialplan, that is free and works. Only need it to work in instances like, "press or say 1..." |
| 22:23.27 | hmmhesays | yes |
| 22:23.30 | hmmhesays | vitelity |
| 22:23.30 | QbY | rrittenhouse.. try http://connect.voicepulse.com -- you can get DIDs, etc, and lots of good examples. |
| 22:23.47 | hmmhesays | theres a whole slew out there |
| 22:24.09 | QbY | Vitelity is equally as good.. although not as many "hold your hand" examples for setup |
| 22:26.39 | rrittenhouse | i was looking at gizmo and buying a few minutes from them (Like $5 worth if possible) |
| 22:26.53 | rrittenhouse | It says you can get 8 hours for $10 I think |
| 22:28.51 | rrittenhouse | Really Im guessing I just need something for outbound calls (I would hook up the POTS line which is technically RoadRunner Digital Phone but i dont have $70 to test an adapter) ;) |
| 22:29.19 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 22:29.20 | rrittenhouse | So im assuming I could get some basic VOIP service to take the outbound calls (inbound too if they provide it) |
| 22:32.31 | *** join/#asterisk d1avlo (n=thi@189.13.219.190) |
| 22:33.15 | d1avlo | Hi, i need help is conf. error in http://paste.milk-it.net/545 |
| 22:33.19 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 22:34.32 | JT | decobbb: what card do you have? |
| 22:49.22 | rrittenhouse | Im assuming it would be better to go with IAX into my asterisk instead of SIP, correct? |
| 22:50.07 | JT | rrittenhouse: nope |
| 22:50.14 | rrittenhouse | oh, ok |
| 22:50.16 | JT | sip should be fine |
| 22:50.25 | JT | and is widely supported |
| 22:50.58 | rrittenhouse | Im just looking through the setup of AsteriskNow and its asking for a service provider and really I dont have an adapter or a serivce provider ;) I wanted to mess with Asterisk with softphones first. |
| 22:51.29 | rrittenhouse | Has anybody here ever tried Gizmo's SIP service for a service provider? |
| 22:51.35 | JT | hrm, asterisknow, i see |
| 22:51.47 | rrittenhouse | Haha. For now. |
| 22:52.18 | rrittenhouse | I want to get acquainted with it before I dive in too much ;) I would love to replace our PBX at work. They are always complaining how expensive it is to upkeep |
| 22:54.55 | JT | ~thebook |
| 22:55.07 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:55.07 | *** join/#asterisk weazahl (n=weazel@adsl-67-65-62-58.dsl.ksc2mo.swbell.net) |
| 22:55.07 | JT | why is it expensive to maintain? |
| 22:55.30 | rrittenhouse | They pay somebody to maintain their current PBX - All i know is who manages it and the type of phones ;) haha |
| 22:55.45 | decobbb | what do i put in zapata.conf for NI-2 signalling ? |
| 22:55.48 | decobbb | national? |
| 22:56.01 | rrittenhouse | Im going to request the information for their PBX and see if they will cough it up so I can try and help them out with a decent pbx. |
| 22:56.10 | rrittenhouse | They just wont be happy about buying new phones :P |
| 22:57.28 | JT | decobbb: yes |
| 22:57.34 | weazahl | why is it that a ulaw call uses ~130kbps not 64k? |
| 22:57.48 | JT | weazahl: should use 85kbit/s |
| 22:57.50 | jpfjp | weazahl: what do you mean? |
| 22:57.52 | JT | not 130 |
| 22:57.59 | decobbb | thank you |
| 22:58.13 | weazahl | i see calls take up 130 + |
| 22:58.20 | JT | weazahl: each way? |
| 22:58.29 | weazahl | combine |
| 22:58.32 | JT | ... |
| 22:58.42 | JT | who measures bandwidth combined? |
| 22:58.45 | weazahl | dsl line is channel bank only |
| 22:58.51 | JT | you measure one way |
| 22:58.52 | weazahl | iptraf |
| 22:58.56 | JT | and it should be 85kbit/s |
| 22:59.13 | JT | go to the correct iptraf screen which shows each direction |
| 22:59.57 | weazahl | yeah i cnat remeber what i saw there |
| 23:00.10 | weazahl | but shouldnt it be 85k total? |
| 23:01.59 | weazahl | would anyone recemend using a codec? channels or quality??? |
| 23:01.59 | *** join/#asterisk shareenergy (n=shareene@195-23-137-126.net.novis.pt) |
| 23:02.25 | JT | weazahl: no, what on earth is wrong with your maths? |
| 23:02.30 | *** join/#asterisk apardo (n=apardo@2001:5c0:9706:0:0:0:0:2) |
| 23:02.36 | JT | weazahl: 85 kilobits per second in each direction. |
| 23:03.30 | weazahl | hey, some details are not completely learned sometimes |
| 23:03.42 | rrittenhouse | Im sure I can set up Asterisk without having any service provider right now, right? Just to mainly test the other features of it. If they do want this at work they can buy the hardware to get it going ;) |
| 23:04.07 | JT | weazahl: how would it be possible that G.711 in SIP would use less bandwidth than the raw g.711 codec? |
| 23:04.15 | shareenergy | anyone can help me with misdn? |
| 23:04.25 | shareenergy | the beronet site is down |
| 23:04.39 | weazahl | ok, so i need to use a codec then |
| 23:04.47 | JT | rrittenhouse: yes you don't need a provider |
| 23:04.58 | rrittenhouse | k i might just do that for now |
| 23:05.00 | weazahl | 4-5 calls will srart to affedt interactivity |
| 23:05.11 | JT | weazahl: what? |
| 23:05.30 | CCFL_Man2 | SwK: y0 |
| 23:05.32 | weazahl | 6mps down 768up |
| 23:05.51 | weazahl | get ofer 600k up and buffers start to clog |
| 23:06.00 | JT | oh ok |
| 23:06.08 | JT | Mbit/s, btw :) |
| 23:06.14 | weazahl | sry |
| 23:06.28 | weazahl | to many herbs... day off you know |
| 23:06.42 | weazahl | what codec would be recomended? |
| 23:06.50 | weazahl | gsm? |
| 23:06.56 | SwK | ? |
| 23:07.04 | JT | with a 768kbit/s upload, why would you be limited to 4 or 5 calls? |
| 23:07.11 | d1avlo | Hi, i need help is conf. error in http://paste.milk-it.net/545 |
| 23:07.50 | weazahl | 600/85+7 |
| 23:07.58 | weazahl | opps =7 |
| 23:08.13 | JT | well that's 7, yeah |
| 23:08.41 | weazahl | at 7 i get lagged out of sequence packets |
| 23:09.05 | JT | must be a dodgy connection :) |
| 23:09.11 | weazahl | anyway to make it shift to gsm at 4 inuse chanels? |
| 23:09.27 | JT | haha, not with asterisk |
| 23:09.35 | mvanbaak | 768 line is going to get you 600 kbps real traffic |
| 23:09.39 | mvanbaak | specially on DSL |
| 23:09.45 | CCFL_Man2 | SwK: i got a cisco mc3810 so i can use it to connect my channel bank to asterisk |
| 23:09.51 | mvanbaak | you have to count the udp and ATM overhead |
| 23:09.58 | JT | only if your provider has a high contention rate :) |
| 23:09.59 | weazahl | well the modem got cooked. they didnt install an a/c and outside temp hit 100F about 120 inside |
| 23:10.08 | JT | yeah, the overhead is nowhere near that high. |
| 23:11.00 | mvanbaak | no, but most of the time the modems buffers will get you stuck at 600 |
| 23:11.16 | SwK | and? |
| 23:11.20 | CCFL_Man2 | SwK: but i'm not sure if the MFT T1 card will connect to my channel bank |
| 23:11.22 | weazahl | so you think my good quality (0 crc errors) should be fine for 8 channels? |
| 23:11.33 | SwK | dunno anything about the mc3810 |
| 23:11.45 | weazahl | i got a dsl2 modem comming |
| 23:11.50 | weazahl | adsl2 |
| 23:11.52 | mvanbaak | SwK: it turned out the buffers were causing too much jitter and variations in the line |
| 23:11.55 | CCFL_Man2 | SwK: cisco says to use the DVM T1 card to connect to pbxs or channel banks |
| 23:12.02 | JT | weazahl: depends on the contention rate of your providers' network |
| 23:12.32 | mvanbaak | as soon as I ditched the default modem and installed a pci atm card I could get close to 512 reliable connection |
| 23:12.39 | weazahl | shorewall can handle buffers, hold p80 traffic for 5060 and 10000:20000 |
| 23:12.57 | weazahl | internal would rock |
| 23:13.04 | Mavvie | oh dear.... problemos majoritus again... |
| 23:13.25 | weazahl | i also can make it roll to the data DSL channel if it maxes out |
| 23:13.33 | mvanbaak | now we run on 1gbit fiber and all trouble is gone |
| 23:13.43 | weazahl | nice |
| 23:13.51 | mvanbaak | yeah |
| 23:13.57 | mvanbaak | 300euro/month for 1gbit fiber |
| 23:14.07 | weazahl | i upgraded a 105 year old hotel |
| 23:14.09 | mvanbaak | very nice deal |
| 23:14.44 | weazahl | VOIP, cat 6, vayer 3 central switch |
| 23:15.13 | weazahl | 32 users now. 60-70 in one year |
| 23:15.56 | weazahl | one 48port poe switch and 3 12/24 poe switchs |
| 23:16.26 | weazahl | 3 hrs of ups to run phones |
| 23:17.54 | weazahl | its cool, http://hotelfrederick.com/ ... im the guy in the black in the slide show. |
| 23:18.49 | `Sean | [7:13pm] <mvanbaak> 300euro/month for 1gbit fiber |
| 23:18.52 | `Sean | wtf where is thatr?> |
| 23:20.06 | weazahl | i want it |
| 23:20.09 | *** join/#asterisk Infested (n=infested@24.148.112.10) |
| 23:20.46 | Weezey | what the shit. Why do none of my prompts generate outbound RTP? |
| 23:21.12 | JT | Weezey: how are you calling the prompts? |
| 23:21.21 | Weezey | VoicmailMain() |
| 23:21.23 | Weezey | Playback() |
| 23:21.26 | Weezey | Background() |
| 23:21.35 | Weezey | nothing generates any audio outbound. |
| 23:21.42 | JT | err, can you pb some of the dialplan involved |
| 23:21.43 | JT | ~pb |
| 23:21.44 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 23:21.52 | mvanbaak | `Sean: .nl |
| 23:22.07 | Weezey | it's all in realtime |
| 23:22.21 | Weezey | I just changed to SVN 1.4, that's the only difference. |
| 23:22.37 | Weezey | I'm going to try rebooting. |
| 23:22.51 | JT | hrm okay |
| 23:23.26 | Weezey | if this doesn't work, I'm going to try unloading the wanrouter (sangoma) and zaptel drivers. |
| 23:23.45 | *** join/#asterisk sacitec (n=tobi@189.129.221.82) |
| 23:23.56 | Weezey | if I do a tcpdump on the box or debug RTP, I don't see anything even attempting to leave. |
| 23:25.38 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 23:26.26 | Weezey | nothing, but it's the only making my case special. |
| 23:27.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:27.55 | JT | aside from the fact you're using SVN |
| 23:28.18 | Weezey | i tried 1.4.10 too, same shit |
| 23:28.46 | Weezey | I'm using SVN-branch-1.4-r79142M |
| 23:28.50 | JT | so what was the last known good version? |
| 23:29.02 | Weezey | 1.2 |
| 23:29.19 | Weezey | (svn something) |
| 23:29.44 | JT | maybe you are using some deprecated dialplan syntax |
| 23:30.03 | Weezey | Hmm, could be. |
| 23:30.24 | Weezey | I tried just punching the extension at VoicemailMail() |
| 23:30.36 | Weezey | and then I tried Answer() then VoicemailMain() |
| 23:30.54 | Weezey | I see the RTP coming in from the devices (iax, sip or gtalk) |
| 23:31.00 | Weezey | but nothing coming out from the * box |
| 23:31.38 | JT | what about Echo? |
| 23:32.02 | Weezey | the echotest? |
| 23:32.39 | JT | does it work? |
| 23:34.00 | Weezey | waiting for the intro to be done playing... |
| 23:34.51 | JT | you're not using trixbox or something are you? |
| 23:35.24 | Weezey | no |
| 23:35.27 | Weezey | no, echo |
| 23:35.42 | JT | what |
| 23:36.09 | Weezey | what what? |
| 23:36.35 | Weezey | Oh, musiconhold works btw |
| 23:36.41 | JT | i have no idea if you are telling me the echo works or not |
| 23:36.42 | Weezey | which is just confusing. |
| 23:36.51 | Weezey | right, bad comma. |
| 23:36.53 | JT | please be less ambiguous |
| 23:36.57 | Weezey | sorry. |
| 23:37.06 | Weezey | No, echo does not work. |
| 23:37.19 | Weezey | no, I'm not using trixbox or some other ez* |
| 23:37.37 | *** join/#asterisk Djeli (n=Djeli@ppp157-243.static.internode.on.net) |
| 23:38.23 | Djeli | Anyone here played with the new Polycom IP550 phones? I'm having a problem getting the Idle image to work (which works on IP501 phones) |
| 23:40.41 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 23:40.55 | Weezey | whoa, I changed musiconhold.conf to be quietmp3 instead of files and now it doesn't work either. |
| 23:41.05 | CCFL_Man2 | SwK: i hear the mc3810 is perfect for a voip gateway |
| 23:45.32 | *** part/#asterisk mtaht4 (n=m@cpe-065-190-150-008.nc.res.rr.com) |
| 23:45.48 | SwK | could be |
| 23:45.50 | SwK | i never used one |
| 23:45.58 | SwK | i stick to IAD or AS series |
| 23:46.20 | JT | what is a mc3810? |
| 23:55.28 | tzanger | where's that jerjer cat |
| 23:56.13 | tzanger | I need to know what the status of h323 and asterisk is these days |
| 23:56.23 | tzanger | there is like a dozen stacks of various stabilities :-) |
| 23:57.41 | *** join/#asterisk ta^3 (n=tacvbo@189.146.191.75) |
| 23:57.44 | *** join/#asterisk tacubo (n=tacvbo@189.146.191.75) |