| 00:15.14 | *** join/#asterisk m_tadeu (~quassel@89-180-187-204.net.novis.pt) |
| 00:19.17 | *** join/#asterisk emsLinux (~emsLinux@190.251.61.192) |
| 00:19.37 | emsLinux | Good night |
| 00:23.44 | emsLinux | I got a question, I have to servers, the first one has Elastix 2, the other one Asterisk 1.8. I configured a SIP trunk betwewn them, they are connected. When i call from an extension in stored in Elastix server to the Asterisk one everything goes ok, but when i try to call to an extension in the Elastix one from the Asterisk one a message came telling me the extension is not available. I think the problem is in the Outbound Routes configuration, i ju |
| 00:23.44 | emsLinux | st used an "1xx" dial pattern in the Elastix Server, but Asterisk 1.8 looks different, i dont know how to do the same in Asterisk 1.8, or maybe i am missing something, can anybody help me? |
| 00:31.23 | *** join/#asterisk m_tadeu (~quassel@89.180.187.204) |
| 00:33.17 | emsLinux | Hello, any help, pls? |
| 00:36.32 | WiretapSeven | anyone here using ISN wanna give me a test dial? |
| 00:41.39 | *** join/#asterisk Chau (~CheeYang@221.72.50.60.brf01-home.tm.net.my) |
| 00:42.21 | Chau | \DSAFEW\, Are you there? |
| 00:42.35 | Chau | \DSAFEW\, I forced to leave the channel last night. |
| 00:42.41 | Chau | \DSAFEW\, I am back now |
| 00:54.00 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 00:56.02 | *** join/#asterisk ccy (~CheeYang@221.72.50.60.brf01-home.tm.net.my) |
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| 01:32.21 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:44.26 | rogersja1 | WiretapSeven: got your ISN all up and running now? |
| 01:45.33 | WiretapSeven | rogersja1, as far as I know, just need someone to dial 2800*1410 to test for me :P |
| 01:45.48 | rogersja1 | standby |
| 01:46.05 | rogersja1 | oh you got 1410⦠i wanted that one :P |
| 01:46.14 | WiretapSeven | why lol? |
| 01:46.33 | WiretapSeven | you can hang up now, my phone is ringing :P |
| 01:46.48 | rogersja1 | well there you go, its working |
| 01:47.05 | rogersja1 | because 1410 is a much nicer number than 1407 |
| 01:47.53 | WiretapSeven | what I wanna be able to do (and I can see you haven't got it working either) is get outbound ISN callerID making sense |
| 01:48.16 | rogersja1 | i haven't bothered trying |
| 01:48.42 | rogersja1 | what does it show as currently? |
| 01:48.59 | WiretapSeven | your name, then "softphone1" |
| 01:52.51 | rogersja1 | cool, then Blink is just setting it as I have it |
| 01:54.15 | rogersja1 | i have overridden it in asterisk, when you're ready i'll try you again |
| 01:54.45 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 01:54.59 | WiretapSeven | whenever you want, I'm just screwing around making a PoE injector |
| 01:55.29 | Scorpio2007 | how to recover from gosub when its sent into invalid extension? for example gosub(blabla|s|1) but blabla doesnt exist? |
| 01:55.44 | WiretapSeven | rogersja1, "Unknown" <softphone1> |
| 01:55.47 | Scorpio2007 | how can i continue to next priority in the dialplan |
| 01:55.56 | ChannelZ | Can you make it work over the interwebs so I can shock deserving people? |
| 02:01.10 | WiretapSeven | ISN Calling 1407 |
| 02:01.14 | WiretapSeven | rogersja1, |
| 02:01.32 | rogersja1 | ugh |
| 02:01.47 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 02:01.52 | WiretapSeven | you wanted the whole number to show not just the calling host I take it |
| 02:02.00 | rogersja1 | can't put asterisk in cid number |
| 02:02.05 | WiretapSeven | yeah |
| 02:02.17 | rogersja1 | going to have to put everything in name i suppose |
| 02:03.27 | rogersja1 | well now that im sidetracked off what i was doing⦠:P |
| 02:09.02 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 02:15.45 | rogersja1 | the hard part is setting callerid on outbound google voice calls ;) |
| 02:15.53 | WiretapSeven | lol |
| 02:16.08 | WiretapSeven | hmm... I wonder if you could get away with using EXT@ISN? |
| 02:16.14 | WiretapSeven | i.e. 2800@1407 |
| 02:16.15 | WiretapSeven | probably not |
| 02:20.50 | WiretapSeven | rogersja1, the cookbook just says to use a valid SIP URI |
| 02:21.55 | rogersja1 | sip uri is fine if you have a full keyboard |
| 02:22.19 | rogersja1 | the whole point of ISN, is for those that cant dial with a full keyboard :P |
| 02:23.11 | WiretapSeven | yeah |
| 02:27.17 | rogersja1 | oh well, back to asterisk and EC2 |
| 02:29.15 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 02:34.36 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 02:36.07 | WiretapSeven | that was weird |
| 02:36.16 | WiretapSeven | somehow callwaiting got disabled for my extn |
| 02:38.50 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 02:43.12 | *** join/#asterisk bintut (~bintut@cm140.kappa8.maxonline.com.sg) |
| 02:43.16 | bintut | waves |
| 02:45.08 | bintut | i need help on sip/tls running on the asterisk v1.8.3.3 from the upstream project.. i'm always getting a 401 unauthorized. i'm using a commercial ssl certificate from comodo's possitivessl |
| 02:47.56 | russellb | we talked about this a while back, i remember ... the 401 unauthorized is not the problem |
| 02:47.59 | russellb | that is the expected response |
| 02:48.14 | russellb | that's asterisk saying "i see your request, now please provide authentication" |
| 02:48.21 | russellb | the phone should be sending another INVITE |
| 02:48.51 | bintut | russellb: that's the problem. i tried bria on iphone4 and snom 300, both cannot proceed |
| 02:50.21 | bintut | is there any specific configuration that I need to add both on asterisk and/or sip client(s)? |
| 02:51.27 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 02:58.34 | bintut | russellb: this is my relevant /etc/asterisk/sip.conf config => http://www.pastie.org/1873842 |
| 03:03.33 | ccy | Does anyone how to check the status of PSTN line ? |
| 03:03.41 | *** join/#asterisk mclaro (~mclaro@host155.200-117-172.telecom.net.ar) |
| 03:07.08 | bintut | dahdi show channel <CHAN_NUMBER> ? |
| 03:09.45 | ccy | bintut, I wish I can get the status via AMI action. |
| 03:10.26 | bintut | i see.. |
| 03:11.36 | ccy | bintut, I have an asterisk appliances that has 10 FXO port, but only 7 FXO has phone line connected. Another 3 ports doesn't has phone lines connected. I want to get the status of these 10 FXO ports. |
| 03:12.11 | bintut | dahdi show channels ? |
| 03:13.09 | ccy | bintut, yes. dahdi show channels return the available fxo I have. but it doesn't report the status of these fxos |
| 03:20.01 | WiretapSeven | woot, DIY PoE injector works |
| 03:20.29 | bintut | ccy: i'm sorry, i'm not aware of that also. :( |
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| 03:37.54 | ccy | How to use GetVar to retrieve Device_State() value? |
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| 03:50.28 | WiretapSeven | ccy, can an FXO even tell it has a line connected until it goes off hook? |
| 03:51.37 | ccy | what is "off hook"? |
| 03:52.03 | WiretapSeven | ~define: off-hook |
| 03:52.11 | WiretapSeven | damnit, that doesn't work in here |
| 03:52.37 | ccy | WiretapSeven, "off hook" means phone is is not plug into the socket? |
| 03:52.44 | bintut | hello..? what am i missing here? => http://www.pastie.org/1873842 |
| 03:55.46 | bintut | i'm testing 2 sip clients here and both of them are getting 401 unauthorized: [1] Bria v1.2.8 build 5312 iPhone Edition and [2] Snom 300 v8.4.31 |
| 04:01.30 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
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| 04:11.24 | WiretapSeven | ccy, off-hook: "Activated, with regard to a telephone set. On a public switched system, a data set that automatically answers is said to go off-hook. Contrast with on-hook." |
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| 04:16.01 | ccy | WiretapSeven, thanks. I use dahdi show channel 1. The hookstate is off hook (phone line connected). dahdi show channel 9 , hookstate is onhook (phone line not plug in) |
| 04:16.25 | ccy | WiretapSeven, this seems to be a correct status I am looking for. |
| 04:16.30 | WiretapSeven | ccy, off hook normally means line in use |
| 04:22.04 | ccy | WiretapSeven, i am monitor my phone lines now. the hookstate is always offhook regardless if the line is in use or idle |
| 04:30.41 | bintut | guys, i found this faq from snom's website => http://is.gd/KB7Lqm |
| 04:31.07 | bintut | since i'm on dns-based tls, does it mean that i have to create a TXT record? |
| 04:32.10 | ccy | WiretapSeven, "dahdi show channel" is console command. how may i use asterisk ami actions to get the hookstate? |
| 04:32.26 | bintut | and also, if my sip phone is connected from my lan that does not have an internal dns, i have to register from the public interface that resolves my domain name? |
| 04:32.36 | bintut | please advice.. |
| 04:37.55 | bintut | sorry, not TXT but SRV record? |
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| 05:01.08 | fauxalliance | gday mzb |
| 05:01.16 | mzb | gday fauxalliance |
| 05:13.05 | bintut | anyone here uses snom 300? |
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| 08:17.17 | serenity | hi |
| 08:17.36 | serenity | can i use asterisk as a MCU for SIP? |
| 08:22.36 | jkroon | MCU? |
| 08:26.47 | serenity | Multipoint Control Unit |
| 08:27.14 | sxpert | wth is that |
| 08:27.50 | serenity | every participant calls sip:foo@bar.com and is then member of a conference |
| 08:28.02 | sxpert | yeah, there's that |
| 08:28.24 | sxpert | you can use the confbridge app |
| 08:29.11 | serenity | ah, mcu's are sometimes called Conference Bridge. Makes sense ;) |
| 08:30.42 | serenity | thanks |
| 08:31.05 | serenity | i have to read lots of documentation |
| 08:31.49 | sxpert | there's an app for that ;) |
| 08:32.24 | serenity | sure, but setting up asterisk in a proper way etc |
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| 11:51.53 | ferdna | getting this error: |
| 11:51.53 | ferdna | [May 7 05:50:18] NOTICE[2398]: chan_sip.c:12357 sip_reg_timeout: -- Registration for 'xxxXXXXXXX@sip.broadvoice.com' timed out, trying again (Attempt #7) |
| 11:53.12 | *** join/#asterisk xoveruk (~rich@193.220.59.2) |
| 11:55.34 | xoveruk | hi |
| 11:56.19 | xoveruk | is it possible to create a custm trixbox installtion that automates the installation onlu asking for ip address |
| 11:58.15 | leifmadsen | xoveruk: please see #trixbox |
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| 12:30.17 | cusco | j edgebox |
| 12:30.19 | cusco | oops |
| 12:47.46 | bintut | anyone here can help me on sips using a domain based commercial ssl on asterisk 1.8.3.3? i am testing on snom 300 v8.4.31 |
| 12:48.12 | bintut | http://www.pastie.org/1873842 |
| 12:49.38 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 12:50.32 | bintut | i'm always getting a 401 unauthorized message. any hint you can share on how to fix this problem? |
| 12:53.10 | DrSbaitso | what is a domain based commercial ssl? i though SSL was port based |
| 12:53.25 | *** join/#asterisk emora (~emora@213.37.32.74.static.user.ono.com) |
| 12:55.06 | bintut | DrSbaitso: basically, i didn't created my own and i bought mine from comodo's possitivessl |
| 12:56.24 | DrSbaitso | i have no idea what Company(tm) has put into their commercial Product(r) |
| 12:58.04 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 12:58.13 | bintut | DrSbaitso: as opposed to what is documented here => https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 12:58.59 | bintut | i use comodo as the ca instead of creating mine as per above tutorial |
| 13:05.51 | bintut | is this correct that the message for the From: and To: lines says something like: sip:361@pbx.domain.com and not sips:361@pbx.domain.com ? please take note of the letter s after sip |
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| 13:34.47 | bintut | DrSbaitso: http://is.gd/KB7Lqm |
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| 14:10.01 | bintut | tsk! my snom 300 is generating this logs => http://www.pastie.org/1874963 |
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| 14:14.22 | applync | Good morning, is it possible to send a 200 OK on the fly from the dial plan? We are forking a call and if one of the endpoints answers, we need to send a 200 OK to another device. |
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| 14:28.40 | ruben23 | hi guys any have idea what is a gatekeeper functionality on h323..? |
| 14:29.45 | bintut | i think this line is the problem => From: "361" <sip:361@pbx.domain.com>;tag=xrgxg6pqvy |
| 14:29.55 | bintut | i think it should be "sips" |
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| 17:04.57 | *** join/#asterisk fragileX (~bill@204.134.45.21) |
| 17:06.32 | fragileX | Im running a new asterisk now 1.6 and Im tring to use zaptel still but is seems that is wants to use dahtti |
| 17:06.49 | fragileX | Im very new to asterisk |
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| 17:13.45 | *** join/#asterisk fragileX (~bill@204.134.45.21) |
| 17:14.39 | fragileX | can i use zaptel instead of dahda |
| 17:14.56 | fragileX | on a 1.6 asterisk now |
| 17:15.44 | fragileX | is anyone here |
| 17:16.11 | *** part/#asterisk fragileX (~bill@204.134.45.21) |
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| 17:35.31 | sezuan | isn't that identical, just renamed? |
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| 19:22.40 | Freeaqingme | one of the agi params is agi_channel, like: agi_channel: SIP/foobar-0000000. What would 'foobar' be in this paste? |
| 19:26.21 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:27.44 | ChannelZ | the device name |
| 19:29.39 | Freeaqingme | ChannelZ, is that the sip user name? |
| 19:30.56 | ChannelZ | well.. you might call it that but technically it's the device name |
| 19:31.36 | ChannelZ | it's the thing [between brackets] in sip.conf |
| 19:31.41 | *** join/#asterisk sizzers (~jerrywilt@c-68-41-174-94.hsd1.mi.comcast.net) |
| 19:32.11 | Freeaqingme | aah, I'm using realtime. Suppose the name entry there is also the device name |
| 19:32.43 | ChannelZ | probably |
| 19:32.58 | Freeaqingme | tnx |
| 19:42.48 | ferdna | what are the required config files for a simple asterisk install? |
| 19:43.11 | ferdna | i have extension.conf, logger.conf, modules.conf, sip.conf |
| 19:45.10 | ChannelZ | Thats probably enough |
| 19:45.26 | ferdna | if i get an error on my asterisk console... will that stop asterisk from working correctly? |
| 19:45.34 | ferdna | error like: [May 7 13:43:55] ERROR[3633]: app_amd.c:429 load_config: Configuration file amd.conf missing. |
| 19:45.49 | ferdna | ChannelZ, ok, thanks =) |
| 19:45.49 | ChannelZ | not necessarily |
| 19:46.04 | ChannelZ | it just means that module or whatever couldn't read its config |
| 19:46.13 | ferdna | ChannelZ, ok... |
| 19:46.36 | ferdna | this is what is stopping me from registering my voip provider: [May 7 13:45:15] NOTICE[3645]: chan_sip.c:12357 sip_reg_timeout: -- Registration for '9154400950@sip.broadvoice.com' timed out, trying again (Attempt #5) |
| 19:47.05 | ChannelZ | well without seeing more, it could be a network problem |
| 19:47.21 | ChannelZ | the packets are never making it to sip.broadvoice.com, or the responses are never making it back |
| 19:47.30 | ferdna | ChannelZ, i can ping, nslookup from bash |
| 19:47.47 | ChannelZ | are you behind NAT or a firewall? |
| 19:47.52 | ferdna | ChannelZ, ipcop |
| 19:48.20 | ChannelZ | well it could be blocking your outgoing SIP packets or not allowing the replies back in |
| 19:48.34 | ferdna | ChannelZ, what port does asterisk use? 5060? |
| 19:48.43 | ChannelZ | yes - UDP |
| 19:48.58 | ChannelZ | then there are a number of ports needed for the audio stream |
| 19:49.16 | ChannelZ | s/stream/streams/ |
| 19:49.55 | ferdna | ChannelZ, do you know what ports are? and if they can be hardcoded into a config file? |
| 19:50.15 | ChannelZ | well for the remote end to send audio to you, it can be anywhere in the port range defined in rtp.conf |
| 19:50.28 | ChannelZ | For you to send audio to the remote end, it's whatever port they request which you have no control of |
| 19:51.19 | ChannelZ | So you pretty much have to allow all UDP out, and then allow whatever range you choose in making sure to port forward if that is your setup as well (IE Asterisk is running on a separate machine behind your firewall, not on the firewall its self) |
| 19:52.14 | ferdna | ChannelZ, i am not using rtp |
| 19:52.24 | ferdna | i have disabled it in sip.conf |
| 19:52.37 | ChannelZ | then what do you expect Asterisk to be doing? |
| 19:53.10 | ferdna | ChannelZ, hehehhe |
| 19:53.11 | ferdna | ok |
| 19:54.06 | ChannelZ | What you probably 'disabled' was forcing Asterisk to be in the middle of the audio streams all the time when it doesn't need to be, but even if your endpoint (SIP phone) is communicating directly with your ITSP, I'm guessing it goes through the same firewall and thus will have the same problem |
| 19:54.28 | ChannelZ | except now you have to forward the audio traffic through the firewall to your phone instead of Asterisk. |
| 19:54.43 | ChannelZ | and those ports would be defined by the phone |
| 19:56.50 | *** join/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 19:57.21 | ferdna | ChannelZ, i have a grandstream bt201 and it just works fine |
| 19:58.35 | ChannelZ | well good |
| 19:59.08 | ferdna | ChannelZ, well... problem is that i cant setup asterisk |
| 20:00.01 | *** join/#asterisk m4xx_ (~m4xx@c-76-19-95-158.hsd1.ct.comcast.net) |
| 20:01.29 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 20:01.50 | ferdna | ChannelZ, how can i know if asterisk is really resolving the dns ? |
| 20:01.57 | ferdna | is there anyway to know? |
| 20:03.25 | *** join/#asterisk fofware (~Fabian@host190.190-31-61.telecom.net.ar) |
| 20:03.45 | fofware | Is there anyone ussing Asterisk.1.8.4-rc3 and Pbx_lua? |
| 20:03.53 | ChannelZ | if the computer it's running on can, asterisk should be |
| 20:04.06 | StefanT | fofware: 1.6.2 and testing pbx_lua atm |
| 20:04.14 | ChannelZ | You said you have a phone, was/is it previously registering directly with your ITSP? |
| 20:04.22 | ferdna | ChannelZ, http://pastebin.com/4jfFHgt6 |
| 20:04.25 | m4xx_ | anyone use the spa50X series ip phones by any chance? |
| 20:04.38 | m4xx_ | by cisco |
| 20:04.40 | ferdna | i XXXXXXX phone# and ips |
| 20:05.38 | ferdna | ChannelZ, yes, it was regisitering directly with the voip provider |
| 20:05.49 | fofware | StefanT: with 1.6.2 I have no problems, but whit 1.8... (1,2,3,4) I have many |
| 20:06.10 | fofware | Maybe I will go back for a time |
| 20:06.23 | ferdna | ChannelZ, does it have to be unregistered... so i can unregister new device? |
| 20:06.23 | fofware | StefanT: thanks any way |
| 20:06.55 | ChannelZ | ferdna: is it still plugged in and configured for the ITSP? if so chances are the return packets your * is sending are getting routed back to the phone |
| 20:07.16 | StefanT | fofware: good to know, I was wondering if I should use 1.8.x |
| 20:09.07 | ChannelZ | gotta go, errands to do. |
| 20:09.19 | StefanT | bye ChannelZ |
| 20:09.59 | StefanT | Anybody here using Sipcall.at ? |
| 20:11.47 | fofware | StefanT: first issue was with contexts, on 1.8.3 pbx_lua don't see context created in extensions.conf, 2° was pickup functions when use it, asterisk freezze, so I did read with 1.8.4-rc3 that not happend, but in this version don't work jabber, and I get error when i do "require "luasql.mysql"" in pbx_lua |
| 20:12.32 | ferdna | ChannelZ, thanks for the help good luck |
| 20:12.40 | StefanT | fofware: Ok, thanks |
| 20:13.40 | m4xx_ | i'm using my devices mac address as the user id. the problem with this is on my cisco phone it show's the user id (it's mac address) next to the line key. anyone know if there's a way to change this behavior? |
| 20:15.24 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 20:16.30 | fofware | you are wellcome, I will back to 1.6.2.18 and lost the new calendar feature :) |
| 20:18.23 | carrar | DAViCAL |
| 20:18.34 | carrar | heh |
| 20:19.11 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:25.04 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 20:25.16 | WiretapSeven | interesting..... wonder if this is a bug in * or FreePBX |
| 20:25.58 | WiretapSeven | when you set an outbound CID including an '@' in the CIDNUM section... it disables CW on all phones until restart |
| 20:36.37 | *** join/#asterisk philippel_mac (~p_lindhei@173-120-199-182.pools.spcsdns.net) |
| 20:36.56 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 20:42.34 | *** join/#asterisk dlublink (~david@75-119-248-158.dsl.teksavvy.com) |
| 20:48.51 | dlublink | I am using asterisk manager to call a Local channel which calls a SIP channel. The call's audio is randomly dropped. If I make the same call without using the local channel, it works fine. I tried core set verbose 1000 and core set debug 1000, but I get no messages. RTP debug shows that asterisk just stops sending RTP packets. |
| 20:49.43 | psykon | m4xx_: You mean it is not as simple as changing the name from the web gui? |
| 20:51.27 | m4xx_ | i didn't see a caption field, perhaps i missed it |
| 20:55.26 | m4xx_ | psykon: brilliant! that lead me to <Short_Name_1_ group="Phone/Line_Key_1"> |
| 20:55.30 | m4xx_ | thank you very much! |
| 21:07.28 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 21:07.54 | nextime | hello all: is there any way to send out of call sip MESSAGEs in asterisk 1.6? |
| 21:11.00 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
| 21:20.02 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
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| 22:09.27 | *** join/#asterisk jpcansa (~JP@201.196.184.142) |
| 22:09.55 | jpcansa | hi, my asterisk-gui "configure hardware" wizard is not detecting my analog card, any idea? using a TDM2400p |
| 22:21.00 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 22:29.45 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 22:32.03 | ferdna | still having problems with my asterisk box |
| 22:32.08 | ferdna | what ports should i have open? |
| 22:32.22 | ferdna | so far i have: rtp:10000-20000 and sip:5060 |
| 22:32.27 | ferdna | forwarded to my box |
| 22:41.46 | ferdna | anyone using broadvoice? |
| 22:42.23 | *** join/#asterisk JerleMinara (~potato@ip68-5-18-63.oc.oc.cox.net) |
| 22:42.51 | JerleMinara | afternoon folks. Anyone mind giving me a hand with using AMI to originate calls on an asterisknow install? ;P |
| 22:43.00 | JerleMinara | (using freepbx to build my dialplans) |
| 23:19.35 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 23:22.23 | nextime | uhmmm, asterisk 1.8.3.3, just upgraded from 1.6.2.9 (both from debian). A csipsimple client ( pjsip based client for android ), on 1.6.2.9 was working, on 1.8.3.3 it can't register on * if i try to use tls transport |
| 23:32.58 | pabelanger | ~collectdebug |
| 23:32.58 | infobot | collectdebug is probably a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 23:33.12 | pabelanger | nextime: PB a debug log from asterisk 1.8 ^ |
| 23:33.45 | JerleMinara | Any suggestion on my stuff, pabelanger? ;P Trying to do an AMI Action: Originate, but just getting "originate failed" and can't root out much more debugging stuff than that? |
| 23:34.32 | pabelanger | JerleMinara: what does your Originate command look like |
| 23:35.02 | JerleMinara | http://www.pastie.org/1876376 |
| 23:35.31 | JerleMinara | and then the to-external context I set up is http://www.pastie.org/1876378 |
| 23:35.43 | JerleMinara | (although obviously I'd rather have something flexible, I was trying to just get the dang thing working for now ;P |
| 23:37.51 | ruben23 | hi guys |
| 23:51.02 | *** join/#asterisk emora (~emora@213.37.32.74.static.user.ono.com) |
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