| 00:04.31 | nextime | mumble |
| 00:04.35 | JerleMinara | i've managed to get things working so if I use "sip/teliax_out/(number)" as the Channel instead of the Exten, I can dial. But swpaping it so that my channel is "SIP/1000" and my exten is "SIP/teliax_out/9499104084" doesn't quite work. |
| 00:04.36 | JerleMinara | any suggestions? ;P |
| 00:04.44 | nextime | it seem that when i send the first REGISTER |
| 00:04.58 | nextime | * answer with a 401 but with a different CSeq |
| 00:05.10 | nextime | so, my client doesn't recognize the answer and drop it |
| 00:05.17 | nextime | and the registration goes in timeout |
| 00:05.52 | nextime | if i use UDP transport, the 401 answer from * has the right CSeq and all is working |
| 00:27.58 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 00:37.46 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 00:39.54 | *** join/#asterisk engrxyz (~puitpyitr@host81-150-217-173.in-addr.btopenworld.com) |
| 00:50.02 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 00:56.37 | ferdna | who owns proxy01.sipphone.com |
| 00:56.56 | ferdna | i also, heard that the stun servers are free and you can use whichever you want |
| 00:59.44 | WiretapSeven | ferdna, do a whois |
| 01:00.34 | IsUp | www.whois.sc/sipphone.com |
| 01:01.50 | ferdna | its google |
| 01:01.52 | ferdna | gizmo |
| 01:01.54 | ferdna | service |
| 01:02.08 | ferdna | ... |
| 01:02.13 | ferdna | where can i get a free sip account? |
| 01:02.22 | ferdna | just like gizmo5 |
| 01:06.58 | *** join/#asterisk fauxalliance (~fauxallia@142.163.128.82) |
| 01:19.08 | *** join/#asterisk bintut (~bintut@cm140.kappa8.maxonline.com.sg) |
| 01:23.27 | bintut | hello..? |
| 01:24.14 | ChannelZ | uhhh.... what am I missing here? A GotoIf is evaluating as GotoIf("SIP/vitelity-in-00000032", ""0"="1"?3,1") and apparently evaluating true... since when does "0" equal "1" ? |
| 01:31.36 | WiretapSeven | ChannelZ, count your " |
| 01:32.19 | WiretapSeven | <PROTECTED> |
| 01:33.39 | ChannelZ | well.. I shoudl say that's the verbose output, not my actual line |
| 01:33.56 | WiretapSeven | oh, well, can you provide the actual line, can't debug with just output |
| 01:34.06 | ChannelZ | the extra quotes are the output of the dialplan |
| 01:34.24 | ChannelZ | It works if I remove the quotes from the channel variable and the evaluation |
| 02:26.16 | *** join/#asterisk dlublink (~david@75-119-248-158.dsl.teksavvy.com) |
| 02:26.28 | dlublink | I am using Asterisk Manager Originate, can I send DTMF when I connect to the remote party ? |
| 02:31.24 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 02:32.34 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 02:39.09 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 03:16.55 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 03:32.36 | ferdna | 606 Not Acceptable |
| 03:32.42 | ferdna | how can i solve this? |
| 03:32.50 | ferdna | what are the parameters that i must change? |
| 03:40.16 | ChannelZ | You asked the other person to do something they didn't want to do. NO MEANS NO! |
| 03:49.09 | ferdna | ChannelZ, was that for me? |
| 03:50.01 | ferdna | i am trying to configure an ekiga account in my asterisk box |
| 03:50.33 | *** join/#asterisk tacloban (~bgiles@c-24-18-11-138.hsd1.wa.comcast.net) |
| 03:51.48 | tacloban | anyone available to help with an issue? I am trying to forward incomming calls to a local extension on my asterisk box. However, the caller just gets an immediate busy signal |
| 03:52.38 | tacloban | the only lines for my incommoing dial plan are: |
| 03:52.43 | tacloban | exten => s,1,Answer() |
| 03:52.47 | tacloban | exten => s,2,Dial(${CISCOATA},20) |
| 03:53.40 | WiretapSeven | well... what is CISCOATA defined as? |
| 03:53.48 | tacloban | CISCOATA=SIP/100 |
| 03:53.59 | tacloban | i have an entry in sip.conf for it |
| 03:54.17 | WiretapSeven | right |
| 03:54.18 | tacloban | i tried both CISCOATA and 100 in the dialplan |
| 03:54.31 | tacloban | the caller still just gets a busy signal |
| 03:54.46 | WiretapSeven | if you were to core set verbose 3, and see what happens in console, then pastebin that I may be able to help |
| 03:55.07 | tacloban | just a sec |
| 03:56.47 | tacloban | so, i did |
| 03:56.57 | tacloban | asterisk -r |
| 03:57.06 | tacloban | then "core set verbose 3" |
| 03:57.16 | WiretapSeven | yep |
| 03:57.28 | WiretapSeven | then hang onto the console while you accept an incoming call |
| 03:57.33 | WiretapSeven | should get output |
| 03:57.35 | tacloban | thats just it |
| 03:57.39 | tacloban | it might be in the asterisk log file? |
| 03:57.49 | WiretapSeven | nope, the output should come onto the console |
| 03:57.50 | tacloban | i am not getting output |
| 03:57.54 | WiretapSeven | you can check the log too |
| 03:57.55 | tacloban | in the console |
| 03:58.02 | WiretapSeven | it should duplicate the console logs you're not getting :P |
| 04:00.41 | tacloban | still trying to find out wth the log is |
| 04:01.03 | tacloban | i see this: astlogdir => /var/log/asterisk |
| 04:01.07 | WiretapSeven | tacloban, /var/log/asterisk/full |
| 04:01.08 | tacloban | but, no log files |
| 04:01.23 | WiretapSeven | are you loading any modules, or is asterisk even running? |
| 04:01.24 | tacloban | no full directory |
| 04:01.28 | tacloban | it is running |
| 04:01.35 | WiretapSeven | full won't be a directory, its the file |
| 04:01.38 | tacloban | do i need to load a logging module |
| 04:01.48 | WiretapSeven | what is your modules.conf? |
| 04:02.07 | tacloban | just this: |
| 04:02.12 | tacloban | autoload=yes |
| 04:02.35 | tacloban | and a few other items that are default |
| 04:02.46 | tacloban | like musiconhold, chan_console |
| 04:04.36 | tacloban | i am testing an outbound call which works fine |
| 04:04.43 | tacloban | but, still no logging on the console or in the log files |
| 04:10.32 | tacloban | okay |
| 04:10.39 | tacloban | so i didnt ever config logger.conf |
| 04:10.53 | tacloban | i am getting stuff in the CLI now |
| 04:11.52 | WiretapSeven | ah, yes, I'm a bit shaky on initial configuration, because mine was performed by freepbx |
| 04:11.56 | tacloban | i get one message |
| 04:12.56 | tacloban | -- Registration for 'user@ekiga.net' timed out, trying again (Attempt #6) |
| 04:13.08 | tacloban | something must be wrong with my sip.conf |
| 04:13.18 | tacloban | i think? |
| 04:13.19 | *** join/#asterisk fauxalliance (~fauxallia@142.163.128.82) |
| 04:13.19 | tacloban | -- Registration for 'user@ekiga.net' timed out, trying again (Attempt #6) |
| 04:14.03 | tacloban | when i do "show sip registry" |
| 04:14.18 | tacloban | ekiga.net:5060 N user 120 Request Sent |
| 04:14.32 | tacloban | so, it must not be registered? |
| 04:14.42 | WiretapSeven | tacloban, it would seem to be the case |
| 04:16.31 | tacloban | i just have this in sip.conf: register => user:password@ekiga.net |
| 04:16.54 | WiretapSeven | the registration timing out might be something more complicated |
| 04:17.04 | WiretapSeven | sip set debug on |
| 04:17.09 | WiretapSeven | and watch the console |
| 04:17.35 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 04:20.21 | tacloban | googling some stuff now |
| 04:20.29 | tacloban | i am getting SIP/2.0 606 Not Acceptable |
| 04:22.14 | tacloban | if i am behind a firewall/NAT/PAT do i have to configure asterisk to send my public ip? |
| 04:25.26 | WiretapMac | yes |
| 04:25.55 | tacloban | can i just use stunaddr? |
| 04:26.32 | WiretapMac | stunaddr just tells it to look for a STUN server |
| 04:26.39 | WiretapMac | I think its 'externaddr' |
| 04:26.49 | WiretapMac | I'm not at a machine with access to my configs at the moment |
| 04:32.07 | WiretapSeven | tacloban, nat=yes, externip=<yourexternaladdr>, localnet=<localsubnet>/<localsubnetmask> |
| 04:36.55 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 04:37.12 | WiretapSeven | rogersja, managed to get a sane CID on ISN |
| 05:03.16 | bintut | wonders who is tacloban since it's a city where he grew up |
| 05:03.27 | bintut | waves.. gtg.. |
| 05:03.28 | tacloban | nope, just been there |
| 05:03.39 | tacloban | no one ever uses it for a nick |
| 05:07.25 | ChannelZ | mmmmm tacos |
| 05:21.47 | rogersja | WiretapSeven, do tell |
| 05:23.33 | WiretapSeven | added this line fater the first line of my ISN outbound |
| 05:23.35 | WiretapSeven | exten => _X.,n,Set(CALLERID(number)=020955${CALLERID(number)}*1410) |
| 05:24.16 | WiretapSeven | which transforms the callerID sent by my outbound route by prefixing "020955", and suffixing "*1410" |
| 05:24.32 | rogersja | interesting |
| 05:24.38 | WiretapSeven | (I also fiddled the outbound config a bit to stop freepbx using my PSTN outbound CID for this route) |
| 05:25.20 | rogersja | i dont use freepbx, so thats not an issue with me |
| 05:25.26 | WiretapSeven | I'm also handling inbound ISN a bit differently now too, so that the above number transforms into a DDI for any valid extension |
| 05:25.37 | WiretapSeven | again hijacking it so freepbx never even gets to touch the call :P |
| 05:25.46 | rogersja | ah |
| 05:25.50 | WiretapSeven | (until it is passed to the from-internal context to dial an inside extension) |
| 05:26.29 | rogersja | i'll have to try it out tomorrow, just landed back home so bed time for me |
| 05:26.34 | WiretapSeven | but yes, it works, I now get nice sane callerid strings that you can even hit return call on :P |
| 05:26.44 | rogersja | cool |
| 05:27.08 | ferdna | softphone for linux? |
| 05:28.29 | rogersja | ferdna: try blink |
| 05:28.40 | ferdna | thats for mac no? |
| 05:29.13 | nellyb | tacloban: i have to use this for a NAT'd PBX: externip=x.x.x.x localnet=192.168.0.0/255.255.255.0 nat=yes |
| 05:29.50 | rogersja | ferdna, they have binaries for debian, and ubuntu |
| 05:30.01 | rogersja | or compile your own from source |
| 05:30.38 | ferdna | i think i will have to compile from source |
| 05:30.40 | ferdna | :D |
| 05:30.49 | rogersja | you'll need Qt |
| 05:31.05 | rogersja | but i'd say blink is byfar the best softphone out there |
| 05:31.12 | rogersja | from what ive seen |
| 05:34.19 | ferdna | i have Qt 7.4.3 |
| 05:34.23 | ferdna | latest one |
| 05:34.25 | ferdna | :) |
| 05:35.48 | ferdna | problem is how to download the src with out darcs |
| 05:54.00 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 05:54.34 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 06:00.58 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 06:01.04 | *** join/#asterisk sideone (~sideone@c-76-110-4-203.hsd1.fl.comcast.net) |
| 06:10.42 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 07:35.21 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 07:35.21 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.3 (2011/04/21), 1.6.2.18 (2011/04/26), 1.4.41 (2011/04/26), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 07:35.45 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:36.26 | AlecTaylor | I'm looking for a web-frontend to join an Asterisk Server with ones microphone. I'm thinking javascript, java, flash or flex. Are there any web-frontends available? |
| 07:37.28 | WiretapSeven | AlecTaylor, spamming your question will not help |
| 07:47.20 | *** join/#asterisk psykon (~psilikon@147-112.96-97.tampabay.res.rr.com) |
| 08:11.44 | WiretapSeven | wonderful |
| 08:11.55 | WiretapSeven | my PSTN SP and my ITSP aren't inter-routing at the moment |
| 08:17.01 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 08:21.52 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 08:29.11 | AlecTaylor | WiretapSeven: errors? |
| 08:32.20 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 08:42.05 | WiretapSeven | AlecTaylor, "ooh nooh, dere es nuting wrrrong" |
| 08:42.15 | WiretapSeven | guy who answered the line barely spoke any english |
| 08:46.20 | *** join/#asterisk AlecTaylor (~AlecTaylo@unaffiliated/alectaylor) |
| 08:47.11 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 08:56.18 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 09:06.29 | AlecTaylor | WiretapSeven: haha |
| 09:06.44 | WiretapSeven | plans are in the works to change ISP |
| 09:14.25 | nellyb | sounds like a good plan! |
| 09:14.47 | nellyb | WiretapSeven, are you in the US/Canada? |
| 09:14.52 | WiretapSeven | NZ |
| 09:15.04 | nellyb | thought you were up late! |
| 09:15.21 | WiretapSeven | lol, its only quarter nine |
| 09:15.43 | nellyb | does DSL work the same way in NZ as in the UK? we get DSL sync and the connect to a radius server via PPPoA. |
| 09:16.44 | WiretapSeven | yep |
| 09:16.47 | nellyb | on some routers I see primary and backup PPPoA connections but BT only allow one PPPoA per line. |
| 09:16.59 | WiretapSeven | here there is only one PVC |
| 09:17.04 | WiretapSeven | and it is _always_ 0/100 |
| 09:17.09 | nellyb | wos a PVC? |
| 09:17.21 | nellyb | virtual circuit> |
| 09:17.35 | WiretapSeven | Permanent Virtual Circuit |
| 09:17.53 | nellyb | ah - thanks |
| 09:18.42 | WiretapSeven | but yes, we have PPPoA on PVC0/100, linespeed is typically unlimited (as fast as the line can sync) |
| 09:18.49 | WiretapSeven | pay by the gigabyte |
| 09:22.40 | nellyb | (sorry this is a bit OT) what sort of speeds do you get? |
| 09:23.47 | WiretapSeven | 15/1 is the average in my area |
| 09:23.49 | WiretapSeven | I see 12/1 |
| 09:24.18 | nellyb | 15 down 1 up? that's good. |
| 09:24.34 | WiretapSeven | yep, we're annex A here, so 1 up is the max |
| 09:25.27 | nellyb | i get 4 down and 1.2 up, but i'm 5.9km from the exchange so it's pretty good. |
| 09:25.39 | WiretapSeven | yeah, that's decent |
| 09:25.53 | WiretapSeven | at work we see 4/.680 |
| 09:26.09 | WiretapSeven | and we're a tech company, go figure :p |
| 09:26.18 | nellyb | a client near birmingham gets 14 down and 2.3 up. amazing. |
| 09:27.26 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 09:27.40 | Dovid | hi all. can anyone make a test call to specififc SIP URI ? |
| 09:29.15 | WiretapSeven | Dovid, if you have an ISN tag for it, sure |
| 09:29.37 | WiretapSeven | otherwise I'm not fighting to type in an alphanumeric URI on my phone :P |
| 09:30.50 | *** join/#asterisk wonderworld (~ww@port-92-201-76-9.dynamic.qsc.de) |
| 09:32.50 | Dovid | WitrtapSeven: 357800800800@109.110.245.78 |
| 09:33.09 | WiretapSeven | Dovid, that is not an ISN dial |
| 09:34.01 | Dovid | ISN == ? |
| 09:35.54 | Dovid | ok. i see what it is |
| 09:36.04 | Dovid | i should create one on freenum ? |
| 09:36.16 | WiretapSeven | http://freenum.org/ http://nerdvittles.com/?p=621 |
| 09:36.34 | WiretapSeven | if you don't have one, it will takea few days to get one |
| 09:36.44 | Dovid | damn, |
| 09:36.46 | Dovid | famn* |
| 09:36.48 | Dovid | damn**** |
| 09:37.10 | WiretapSeven | mine took me a week for the IANA response, and 48 hours for the freenum response |
| 09:37.15 | Dovid | my issue is that all of my IP's are registerd in my firewall. wanted to see if the our side access works |
| 09:37.26 | WiretapSeven | ah, I see |
| 09:37.42 | WiretapSeven | get someone local to call your DID? |
| 09:37.53 | Dovid | it should accpet anything to 357800 |
| 09:38.00 | Dovid | i want to give Free Toll Free to Cyprus |
| 09:38.14 | Dovid | i want to see if people can just send to my procy |
| 09:38.17 | Dovid | proxy |
| 09:39.37 | Dovid | anyone else out there ? |
| 09:48.13 | WiretapSeven | Dovid, not too many have URI dialling set up I don't think |
| 09:49.02 | Dovid | but u can do it in Asteirsk |
| 09:49.09 | Dovid | thats how i test |
| 10:30.21 | *** join/#asterisk manji (~manjiki@ix.ceid.upatras.gr) |
| 10:40.15 | *** join/#asterisk manji (~manjiki@ix.ceid.upatras.gr) |
| 11:14.08 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 11:15.41 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 11:16.56 | RypPn | congested to SIP/357800800800@109.110.245.78 |
| 11:19.51 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 11:30.34 | Dovid | RypPn: Still here ? |
| 11:30.58 | *** join/#asterisk jkroon (~jkroon@dsl-241-248-179.telkomadsl.co.za) |
| 11:31.07 | RypPn | Dovid yup |
| 11:31.21 | Dovid | Ok. let me do a trace |
| 11:32.23 | RypPn | ready? |
| 11:33.03 | Dovid | argh. i seem to have soe kind of loop. or something wrong here. |
| 11:34.26 | jkroon | Dovid, don't we all :p |
| 11:37.13 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 11:40.21 | Dovid | looks like my gateway went nuts. argh !!! |
| 11:54.35 | Dovid | jkroon: Small Typo. Can you try 35780000008@109.110.256.78 ? |
| 11:55.06 | jkroon | hi guys, when a SIP UA sends me a 302 it's possible to pick up the channel via CALLERID(RDNIS), does anybody know how I can pick up the IP that sent me the 302? |
| 11:55.20 | jkroon | Dovid, actually no I can't, not at the moment anyway. |
| 11:55.55 | Dovid | ok. let me know if and when u can |
| 11:56.26 | jkroon | Dovid, i don't even know what the problem there is you're after, but give me 5 anyway. |
| 11:57.05 | nellyb | David: Looks like SIP external is getting through: 109.110.245.78:5060 | PBX_MANAGER | Grandstream phone |
| 11:57.42 | Dovid | good. thats what I want |
| 11:57.48 | jkroon | <PROTECTED> |
| 11:57.52 | Dovid | yup |
| 11:57.58 | Dovid | only 357800 should |
| 11:58.00 | Dovid | nothing else |
| 11:58.25 | jkroon | confused. |
| 11:58.44 | Dovid | I am a liscenced carrier in Cyprus. I get compensated for Toll Free Minutes |
| 11:58.48 | Dovid | the same way people do it in the US |
| 11:59.01 | jkroon | anyway, after re-entering the dialplan on a SIP 302 redirect I need to pick up the IP that initiated the redirect - is this possible? |
| 11:59.01 | Dovid | I do it in the US as well but we are still testing there |
| 11:59.20 | jkroon | in ZA |
| 11:59.31 | AlecTaylor | What advantage is there to an ISN? |
| 11:59.37 | Dovid | if you have Cyprus TF traffic send it my way ;) |
| 11:59.38 | jkroon | where the recipient of a toll free call picks up the bill :p |
| 11:59.52 | Dovid | jkroon: Same here. since I am a carrier I get a kick back |
| 12:00.26 | jkroon | Dovid, ah ok. i sometimes want to point a bazooka at the carriers here. |
| 12:00.51 | Dovid | lol |
| 12:01.09 | jkroon | it's not funny :p |
| 12:01.39 | jkroon | ok, so no one knows how to find the IP of the device that sent the 302 request I take it? |
| 12:05.45 | jkroon | hmm, let's say I want to CUT up a variables that may contain one of more -'s, but I all fields except the last one, ie, a-b-c should become a-b - CUT() can do this if I know the number of fields, ideas? |
| 12:39.00 | Gugge | jkroon: theres a function that gives you the number |
| 12:42.52 | jkroon | Gugge, i've now read the entire list from core show functions and i haven't seen it. mind sharing? |
| 12:42.56 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 12:49.42 | Gugge | FIELDQTY |
| 12:50.03 | jkroon | thanks! |
| 12:50.15 | Gugge | http://www.voip-info.org/wiki/view/Asterisk+func+cut <- the first "see also" function :) |
| 12:50.50 | jkroon | strange, core show function CUT has "Not available" in the [See Also] |
| 12:50.56 | *** join/#asterisk manji (~manjiki@ix.ceid.upatras.gr) |
| 12:51.16 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc7-pool12-2-0-cust219.15-1.cable.virginmedia.com) |
| 12:51.31 | MarKsaitis | is this guide enough https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages to start using asterisk1.8 with srtp? |
| 12:51.44 | MarKsaitis | or do I need to compile asterisk on my own? |
| 12:51.51 | MarKsaitis | or what? |
| 12:55.26 | EmleyMoor | Is there a way, with Asterisk's help, to call a "short" French telephone number? (i.e. a non-geographical one having only 4 digits) |
| 12:56.59 | MarKsaitis | do asterisk 1.8 packages on debian come with full srtp support or not? |
| 12:58.14 | MarKsaitis | how do i check it |
| 12:58.27 | Freeaqingme | install it |
| 12:58.28 | Freeaqingme | check it |
| 12:58.37 | MarKsaitis | I have installed it just nw |
| 12:59.01 | MarKsaitis | last year i compiled myself asterisk with srtp, but it all had some problemds |
| 12:59.10 | MarKsaitis | I am trying it all again just now |
| 12:59.24 | MarKsaitis | how do i check for it? |
| 13:01.17 | jkroon | EmleyMoor, probably yes. |
| 13:01.30 | jkroon | perhaps you need to explain the requirement a bit? |
| 13:01.59 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 13:02.11 | EmleyMoor | I need to call "3235" - but I am in England. |
| 13:02.29 | jkroon | and you need to land that in france? |
| 13:02.43 | jkroon | do you have an asterisk box connected to the france pstn somewhere? |
| 13:02.46 | MarKsaitis | so how do I check if my debian installation of asterisk 1.8 supports srtp? i have followed official deigium asterisk guide on debian |
| 13:03.05 | EmleyMoor | jkroon: No - my only Asterisk box is in England too |
| 13:03.09 | jkroon | MarKsaitis, good question. |
| 13:03.30 | MarKsaitis | jkroon, I know. It is so hard to go and progress with these things |
| 13:03.34 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 13:03.54 | MarKsaitis | asterisk page for that debian package doesnt say whats inside whatsoever |
| 13:03.54 | jkroon | EmleyMoor, unless you know of a way to call that number from outside of france you're stuffed. |
| 13:04.16 | jkroon | MarKsaitis, i'm on ast 1.6.2, and it has an rtp command. |
| 13:04.25 | EmleyMoor | I'm looking to see if there is a numero vert that does the same thing |
| 13:04.27 | MarKsaitis | no, I need srtp |
| 13:04.31 | jkroon | perhaps 1.8 has something like rtp show ??? that can show what versions is available? |
| 13:04.32 | MarKsaitis | and its a feature |
| 13:04.43 | MarKsaitis | secure rtp |
| 13:04.55 | MarKsaitis | which uses libsrtp |
| 13:05.17 | jkroon | ok, then in /usr/lib/asterisk/modules, use ldd to check whether there is anything that links with libsrtp |
| 13:05.30 | jkroon | for i in /usr/lib/asterisk/modules; do ldd $i; done | grep libsrtp |
| 13:05.38 | jkroon | for i in /usr/lib/asterisk/modules/*; do ldd $i; done | grep libsrtp |
| 13:07.12 | MarKsaitis | jkroon, returns nothing |
| 13:07.34 | jkroon | then the answer to your question is no, i don't think you've got srtp support available. |
| 13:08.02 | *** join/#asterisk jdoe (jdoe@falseprophet.ca) |
| 13:08.03 | jkroon | knowing debian though there is probably just an extra package you need to install to get it, or failing that you're back to compiling. |
| 13:08.11 | jkroon | or file a bug against the package manager and wait. |
| 13:08.15 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 13:08.24 | MarKsaitis | this page says that asterisk 1.8 has native support for srtp http://www.voip-info.org/wiki/view/Asterisk+SRTP |
| 13:08.47 | MarKsaitis | this just sucks so much |
| 13:08.53 | jkroon | perhaps just try using srtp from a phone and see if it works? |
| 13:09.01 | jkroon | sometimes the easiest solution is to just try. |
| 13:09.26 | MarKsaitis | jkroon, trying and using srtp from phon is more complicated than finding if it actually is in my installation :) |
| 13:09.30 | MarKsaitis | I know |
| 13:09.53 | MarKsaitis | this is a fresh install, so I do not have anything to start testing it with phones just as of yet |
| 13:09.56 | jkroon | no it's not. even my sucky grandstream has an option "srtp support: off, optional, required" |
| 13:10.29 | MarKsaitis | I only can test with some softphones here |
| 13:12.56 | MarKsaitis | All I want is just a damn asterisk 1.8 package installed on my system and srtp working |
| 13:13.04 | MarKsaitis | and then test it with couple of softphones |
| 13:13.21 | MarKsaitis | <PROTECTED> |
| 13:14.04 | jkroon | rolls eyes |
| 13:14.20 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 13:17.15 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc7-pool12-2-0-cust219.15-1.cable.virginmedia.com) |
| 13:17.26 | MarKsaitis | ;] |
| 13:17.52 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 13:18.50 | MarKsaitis | so does anybody KNOW WHETHER OFFICIAL ASTERISK 1.8 DEBIAN SQUEEZE PACKAGE SUPPORTS SRTP STR8 OUT OF THE BOX??? |
| 13:21.32 | MarKsaitis | Qwell, |
| 13:31.01 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 13:51.04 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:56.49 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:22.23 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 14:27.50 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 14:39.42 | Freeaqingme | MarKsaitis, use of capitals on irc is considered shouting |
| 14:39.51 | Freeaqingme | and therefore considered offensive and likely to be ignored |
| 14:40.00 | MarKsaitis | Freeaqingme, and I was shouting |
| 14:40.09 | Freeaqingme | not helping ;) |
| 14:40.17 | MarKsaitis | because otherwise nobody hears anything :) |
| 14:40.19 | MarKsaitis | thanx :) |
| 14:40.27 | Freeaqingme | if it's really, really needed and you cant figure it out, just contact digium's sales/support dept |
| 14:40.39 | MarKsaitis | u usually shout if everybody is far away from you |
| 14:41.16 | MarKsaitis | well I know how to get srtp working, but it needs to compile everything |
| 14:41.19 | Freeaqingme | my screen is only 2-3 ft away from me, so that's not far |
| 14:41.23 | Freeaqingme | then compile it? |
| 14:41.29 | MarKsaitis | I just want to know if all that stuff is done in asterisk 1.8 |
| 14:41.36 | MarKsaitis | I dont want to compile it for simplicity |
| 14:41.48 | MarKsaitis | but if there will be no other way, I will need to |
| 14:41.55 | Freeaqingme | if you'd had compiled it you would have been done already |
| 14:42.23 | MarKsaitis | Freeaqingme, well, dunno where ur scren is, but it took you hours to hear my message. Imagine if I wasnt shouting then :) |
| 14:42.37 | MarKsaitis | Freeaqingme, well, I did so half a year ago |
| 14:42.48 | MarKsaitis | but most softphones I tested it all sucked really bad |
| 14:42.54 | MarKsaitis | this is another attempt now |
| 14:43.04 | MarKsaitis | si, I am not done |
| 14:43.08 | Freeaqingme | then again, file an issue report, file a patch, or contact digium's commercial dept |
| 14:44.12 | *** join/#asterisk ashd (~ashleyd@conres.c-r.org) |
| 14:44.57 | MarKsaitis | I hate filing this shit, never do. I just want an answer whether it is in the box or not :) but yes, if nobody will answer this today, I will need to do that or smth else |
| 14:52.00 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 14:55.57 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:58.02 | *** join/#asterisk war9407 (war@c-71-62-61-45.hsd1.va.comcast.net) |
| 15:00.34 | *** join/#asterisk jcook_5xdata (~jcook_5xd@c-98-243-8-24.hsd1.mi.comcast.net) |
| 15:01.39 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 15:35.09 | jcook_5xdata | I have asterisk 1.4 polycom 601 sip.ld 3.1.7. I am trying to get presence to work. I follow http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup. But I still do not get any 'sip show subscriptions' Any thoughts? |
| 15:36.27 | jcook_5xdata | And I see no watchers when I do 'core show hints |
| 15:36.27 | jcook_5xdata | ' |
| 15:37.49 | bdfoster | sip show peers |
| 15:38.00 | bdfoster | sip show registry |
| 15:38.10 | bdfoster | idk lol |
| 15:39.28 | jcook_5xdata | sip show peers give http://pastebin.com/LHp9qaxb |
| 15:39.58 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-231.ips.direcpath.com) |
| 15:40.23 | jkroon | similar to SIPPEER(peername,chanvar[varname]), is there something to get the variables off of IAX user definitions? |
| 15:43.17 | jkroon | nope, that sounds like a resounding NO and chan_iax.c concurs. |
| 15:53.14 | *** join/#asterisk davlefou (~david@41.225.55.153) |
| 15:59.59 | *** join/#asterisk sourcode (~code@ppp-61-90-39-190.revip.asianet.co.th) |
| 16:01.41 | *** join/#asterisk davlefou (~david@41.227.58.42) |
| 16:08.38 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 16:10.11 | *** join/#asterisk davlefou (~david@41.225.83.53) |
| 16:10.13 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 16:25.46 | *** join/#asterisk davlefou (~david@41.225.21.247) |
| 16:47.53 | *** join/#asterisk davlefou (~david@196.203.27.2) |
| 16:54.30 | jcook_5xdata | I have asterisk 1.4 polycom 601 sip.ld 3.1.7. I am trying to get presence to work. I follow http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup. But I still do not get any 'sip show subscriptions' Any thoughts?here is a sip debug of subscription http://pastebin.com/a6k53G0W |
| 16:58.21 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 17:11.08 | *** join/#asterisk davlefou (~david@41.225.127.238) |
| 17:25.36 | *** join/#asterisk davlefou (~david@41.225.223.230) |
| 17:45.50 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 17:45.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:47.04 | *** join/#asterisk davlefou (~david@41.227.20.84) |
| 17:56.02 | *** join/#asterisk monzsca (~monzsca@74-139-22-98.dhcp.insightbb.com) |
| 18:09.33 | *** join/#asterisk davlefou (~david@41.225.90.12) |
| 18:35.18 | *** join/#asterisk davlefou (~david@41.225.102.114) |
| 18:38.45 | *** join/#asterisk mclaro (~mclaro@201.250.102.16) |
| 18:44.42 | nellyb | anyone providing hosted voip in fresno, ca? chatting with a guy on linkedin about it - he's got a bad taste in his mouth after a poor implemetation (also posted on #freepbx) |
| 18:44.59 | *** join/#asterisk davlefou (~david@41.227.63.31) |
| 18:52.26 | *** join/#asterisk davlefou (~david@41.225.254.231) |
| 18:52.46 | \DSAFEW\ | nellyb, I'm pretty sure dslreports has voip in it's local ISP section, but that might just be consumer telephony, hit the phonebook too |
| 18:57.21 | nellyb | thanks DSAFEW! |
| 19:01.18 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1176003507.dsl.bell.ca) |
| 19:10.28 | *** join/#asterisk jcook_5xdata (~jcook_5xd@c-98-243-8-24.hsd1.mi.comcast.net) |
| 19:11.57 | *** join/#asterisk davlefou (~david@41.227.38.221) |
| 19:12.21 | jcook_5xdata | I have asterisk 1.4 polycom 601 sip.ld 3.1.7. I am trying to get presence to work. I follow http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup. But I still do not get any 'sip show subscriptions' Any thoughts?here is a sip debug of subscription http://pastebin.com/a6k53G0W |
| 19:13.12 | sxpert | hmm, you're looking for #freeswitch I guess |
| 19:14.18 | sxpert | alternatively, you should be looking here http://www.voip-info.org/wiki/view/Asterisk+presence |
| 19:18.43 | jcook_5xdata | try that one as will. From the sip debug it have a tried time look up hints |
| 19:18.59 | jcook_5xdata | I did upgrade to 1.6 to see that helps |
| 19:19.06 | jcook_5xdata | did not |
| 19:19.08 | jcook_5xdata | :( |
| 19:19.52 | jcook_5xdata | From what I read hints was added to 1.4 + to helps this problem |
| 19:20.53 | sxpert | attempting to apply a freeswitch doc to as asterisk install has a very low chance of being successful ;) |
| 19:23.01 | jcook_5xdata | I seen this one http://www.voip-forum.com/asterisk/2008-01/sip-subscriptions/ |
| 19:23.30 | Gugge | jcook_5xdata: did you add the hint to the extension in the dialplan |
| 19:25.17 | jcook_5xdata | nope, I will try that thou... |
| 19:25.27 | Gugge | ... |
| 19:25.34 | Gugge | how do you expect it to work then? :) |
| 19:25.52 | Gugge | if you dont configure what device you want status from on the extension |
| 19:26.55 | jcook_5xdata | like I said. I have read that 1.4+ did not need it becouse of hints "core show hints" |
| 19:27.11 | Gugge | where did you read that? |
| 19:27.23 | jcook_5xdata | but yea it my not work right |
| 19:27.52 | Gugge | how would asterisk know what sip device to show status for on the extension if you dont tell it? |
| 19:29.02 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 19:31.27 | jcook_5xdata | couple of forum post.. |
| 19:31.52 | jcook_5xdata | OK so I added my test ext's |
| 19:32.25 | *** join/#asterisk davlefou (~david@41.225.211.50) |
| 19:32.49 | jcook_5xdata | i my sip debug Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO it allowed |
| 19:35.11 | *** join/#asterisk v1s (~v1s@124.6.181.167) |
| 19:36.55 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 19:39.01 | v1s | I have a server with multiple IP address 2.2.2.X. I have sip.conf set to bindaddr=0.0.0.0 I am using vpn for server to server communication which is using say ip 3.3.3.X they are working fine. And the box is working find as an sip client to my providers. But if I try to connect as a client to a ip it wont work unless I go through all the ips and find out which one its oh is there way Ican set which port it should listen/transmit on and still be able to do it on |
| 19:39.28 | jcook_5xdata | Gugge, thank you for Your help I missed having me look. Anyway Ploycom or asterisk is looking for [device-hints] for the hints. out put for sip debug - Looking for 101 in device-hints added that I was Golden |
| 19:39.32 | v1s | port = ip |
| 19:49.03 | sezuan | is it possible to react on incoming skype messages with chan_skype? |
| 19:49.39 | sezuan | like forwarding the messages to another IM account. |
| 20:01.27 | *** join/#asterisk Egyptian[Laptop] (~marafa@adsl-99-159-248-253.dsl.wotnoh.sbcglobal.net) |
| 20:03.39 | Egyptian[Laptop] | hi - can someone point me to documentation to building a gsm to voip gateway (and vice versa!) ? |
| 20:04.39 | sxpert | Egyptian[Laptop]: which part of gsm ? |
| 20:04.49 | *** join/#asterisk lpmusic (~dballenge@denetronllc-1-pt.tunnel.tserv3.fmt2.ipv6.he.net) |
| 20:04.56 | Egyptian[Laptop] | sxpert: errr ??? |
| 20:04.56 | sxpert | you want to act as a series of phones, or you want to build a base station ? |
| 20:05.12 | Egyptian[Laptop] | sxpert: no no .. for personal use only - i am not a telco |
| 20:05.21 | sxpert | hah |
| 20:05.42 | sxpert | so you need a gsm PCIe card that you put in your PC... done |
| 20:06.06 | Egyptian[Laptop] | sxpert: ok - what card ? and what software ? |
| 20:06.32 | sxpert | everything is here http://www.voip-info.org/wiki/view/VOIP+GSM+Gateways |
| 20:06.44 | Egyptian[Laptop] | sxpert: ok thanks :) |
| 20:08.16 | Egyptian[Laptop] | wow! thats a lot of hardware to pick from .. anyways what about software? i mean i am assuming asterisk but how about configuration ? |
| 20:10.25 | sxpert | some of those do the gsm<=>sip and present an ethernet interface. they show up as sip endpoints |
| 20:12.24 | Egyptian[Laptop] | sxpert: i do not understand sip terminology all i know is i have an idea in my head of what i want to do and it requires hardware and software and some personal hardwork :) |
| 20:12.54 | Egyptian[Laptop] | so is the feature in the hardware i am looking for called "sip endpoint" ? |
| 20:12.57 | sxpert | basically they present themselves as a phone |
| 20:13.16 | Egyptian[Laptop] | i already have the gsm sim card |
| 20:16.17 | sxpert | yeah, you plug the sim in the device, and it assumes the identity of the sim just like a phone |
| 20:16.34 | sxpert | you probably want to disable the pin code |
| 20:23.22 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 20:24.49 | Egyptian[Laptop] | i can do that ? |
| 20:24.56 | sxpert | yeah |
| 20:25.36 | Egyptian[Laptop] | ok .. why would i want to do that? and seriously if you want to ..just point me to the documentation .. i can read ;) |
| 20:25.45 | Dovid | Egyptian[Laptop]: Hello neighbor ;) |
| 20:25.57 | sxpert | Egyptian[Laptop]: I just did ;) |
| 20:26.10 | Egyptian[Laptop] | its all in the link above? thats hardware only .no? |
| 20:26.16 | Egyptian[Laptop] | hello Dovid |
| 20:26.16 | sxpert | Egyptian[Laptop]: you need a sim card for the sip=>gsm gateway to work |
| 20:26.45 | sxpert | as it needs a subscription to access the telco's network |
| 20:27.32 | Egyptian[Laptop] | sxpert: thats a taken |
| 20:28.13 | sxpert | the actual term is "a given" |
| 20:29.51 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:30.30 | Egyptian[Laptop] | taken for granted then |
| 20:32.55 | sxpert | then you should be all set |
| 20:33.04 | Egyptian[Laptop] | may i recap? all i need is hardware called sip endpoint from the above list and a gsm sim. |
| 20:33.13 | Egyptian[Laptop] | what about asterisk configuration ? |
| 20:33.14 | sxpert | get one of those gsm=> sip/ethernet gateways, and insert sim. done |
| 20:33.29 | sxpert | the gateway is viewed as a sip phone |
| 20:34.17 | sxpert | so you configure your asterisk to accept it as a sip hone |
| 20:34.20 | sxpert | phone |
| 20:34.21 | sxpert | done |
| 20:34.47 | Egyptian[Laptop] | ok .. sounds simple enough |
| 20:36.50 | *** join/#asterisk manji (~manjiki@ppp-2-84-10-110.home.otenet.gr) |
| 20:45.24 | *** join/#asterisk squidly (~squidly@HoodLUG/member/squidly) |
| 20:45.47 | squidly | hello I'm having issues with trixbox and outbound calls. I keep getting "all circuits are busy" when I try to call out |
| 20:49.14 | squidly | I'm not sure where to look to debug this. I'm able to get incomeing calls with out issues |
| 20:50.01 | Chainsaw | squidly: Well these outbound calls are going somewhere. |
| 20:50.50 | Chainsaw | squidly: I'm assuming you plugged into a telephone network somehow. It would be helpful to know what device you are using for that. |
| 20:51.49 | squidly | I'm using broadvoice for my outbound peer |
| 20:52.23 | Chainsaw | Right. Have you contacted them? |
| 20:52.59 | squidly | well when I was using my wifi phone configured to go directly out to broadvoice it was working fine. |
| 20:54.24 | squidly | and I can get incomeing calls with an issue |
| 20:56.06 | Chainsaw | squidly: Then I would suggest that you compare configs between the software that works (WiFi phone) and the software that does not (Trixbox). |
| 20:56.33 | Chainsaw | squidly: Alternatively you could use a packet capture tool like wireshark to analyse the conversations between your software and BroadVoice as it happens. |
| 21:18.37 | squidly | the setting are the same as far as I can tell. |
| 21:18.42 | squidly | I can get incomeing calls no problem |
| 21:36.54 | Chainsaw | squidly: Then it is time to move on to the wireshark part of the exercise. |
| 21:53.11 | *** join/#asterisk Egyptian[Laptop] (~marafa@adsl-99-159-248-253.dsl.wotnoh.sbcglobal.net) |
| 22:16.32 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:53.33 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 23:11.02 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 23:17.28 | *** join/#asterisk Egyptian[Laptop] (~marafa@adsl-99-159-248-253.dsl.wotnoh.sbcglobal.net) |
| 23:33.05 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 23:47.57 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-66-67-121-128.rochester.res.rr.com) |