| 00:06.07 | *** join/#asterisk edguy3 (~colloquyu@c-98-221-27-224.hsd1.nj.comcast.net) |
| 00:07.41 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 00:10.02 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 00:32.38 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:43.41 | *** join/#asterisk bmg505 (~leon@196-209-84-63.dynamic.isadsl.co.za) |
| 00:44.26 | dijib | soup y'all |
| 00:46.37 | [TK]D-Fender | NO SOUP FOR YOU! |
| 00:47.20 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 01:05.54 | *** join/#asterisk nzkiwi1 (~michaelnz@ip210-48-102-29.nettrust.net.nz) |
| 01:06.47 | nzkiwi1 | hi. I am having problems with asterisk and TLS |
| 01:07.06 | nzkiwi1 | I have now tried Blink client and that does not work either |
| 01:07.19 | nzkiwi1 | however my PBX does work when talking to another asterisk box |
| 01:07.26 | nzkiwi1 | with TLS |
| 01:07.54 | nzkiwi1 | tcptls.c:235 handle_tcptls_connection: FILE * open failed |
| 01:08.01 | nzkiwi1 | anyone recognise this? |
| 01:12.08 | dijib | ive always soup'in' |
| 01:12.14 | dijib | im |
| 01:12.16 | dijib | i am |
| 01:12.31 | dijib | why is SeRi not in here |
| 01:12.38 | dijib | home life.. hope he's been working on it |
| 01:15.36 | nzkiwi1 | no one? are you just here for yourselves? |
| 01:24.03 | dym | nzkiwi1: have you tried googling before? |
| 01:24.12 | dym | is this interconnection between two * boxes? |
| 01:24.26 | dym | what exactly are you trying to do, and where do you get this error? |
| 01:27.15 | nzkiwi1 | I have spent hours on info obtained from Google |
| 01:27.30 | nzkiwi1 | I have done everything with the official docs |
| 01:27.42 | nzkiwi1 | the two are directly connected |
| 01:27.49 | nzkiwi1 | across a lan |
| 01:27.53 | dym | 2 asterisk servers? |
| 01:28.07 | nzkiwi1 | the asterisk server is connected across the net |
| 01:28.12 | nzkiwi1 | both on public IPs |
| 01:28.18 | nzkiwi1 | that works. Nothing else does |
| 01:28.36 | dym | nzkiwi1: you're not providing essential information |
| 01:28.41 | dym | Are we talking about 2 asterisk servers? |
| 01:28.44 | nzkiwi1 | a yealink T28p phone or a blick client. both work with udp but not tls |
| 01:28.49 | dym | Or a Client => server Connection? |
| 01:29.25 | nzkiwi1 | I have the blick client on my laptop which is directly connected to the asterisk server |
| 01:29.31 | nzkiwi1 | blick |
| 01:29.36 | nzkiwi1 | blink sorry |
| 01:30.26 | dym | ah |
| 01:30.27 | nzkiwi1 | I can make calls from the voip phone or the blink client but not receive. sip show peer xxx shows them as unreachable |
| 01:30.37 | dym | can you reach the server? |
| 01:30.45 | nzkiwi1 | yes |
| 01:30.50 | nzkiwi1 | can ping both ways |
| 01:30.54 | dym | right |
| 01:31.04 | dym | you connect using a client - what does the asterisk CLI say? |
| 01:31.38 | nzkiwi1 | tcptls.c:235 handle_tcptls_connection: FILE * open failed |
| 01:31.45 | dym | is this the asterisk cli? |
| 01:31.50 | nzkiwi1 | at the moment it is giving me this message every minute or so |
| 01:31.55 | dym | and a direct reaction to your connect? |
| 01:32.05 | nzkiwi1 | it is the cli |
| 01:32.59 | dym | do you have a directive named "tlscipher=DES-CBC3-SHA" in your sip.conf? |
| 01:33.08 | nzkiwi1 | tlscipher=ALL |
| 01:33.11 | dym | (poking in the dark here) |
| 01:33.19 | dym | comment that please (just to try) |
| 01:33.29 | nzkiwi1 | ok brb |
| 01:33.30 | dym | then retry |
| 01:34.24 | nzkiwi1 | tlscipher = DES-CBC3-SHA |
| 01:34.38 | nzkiwi1 | I have also set tlsdontverifyserver = yes |
| 01:34.38 | nzkiwi1 | tlsclientmethod = tlsv1 |
| 01:34.38 | dym | What? |
| 01:34.49 | dym | Whats with the first one you just quoted - is that there? |
| 01:36.25 | nzkiwi1 | tlsdontverifyserver = yes |
| 01:36.29 | nzkiwi1 | and no change |
| 01:36.54 | nzkiwi1 | still getting that message and cant call to client (says unrecahable) |
| 01:37.59 | dym | well |
| 01:38.04 | dym | then the client isnt even registered |
| 01:38.09 | dym | you should first get that sorted out |
| 01:38.14 | dym | why is the client not registered |
| 01:38.23 | nzkiwi1 | it is registered |
| 01:38.26 | nzkiwi1 | it says so |
| 01:39.00 | nzkiwi1 | same with the yealink phone. it registers but is unreachable |
| 01:39.57 | nzkiwi1 | I work with networking and servers all the time and this is one of the most confounding issues I have ever had |
| 01:40.51 | nzkiwi1 | it does not make sense and noone seems to know the answer (tho doing a search on the messages shows others have had these issues back as far as 2008/2009) |
| 01:41.32 | nzkiwi1 | this pbx can talk tls fine to the ITSP who also runs asterisk. everything else nup. only UDP |
| 01:50.37 | dym | how do you know the phone is registered? |
| 01:52.06 | nzkiwi1 | it says "registered" and in sip show peer it shows the ip it is on |
| 01:52.12 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-228-059.pools.arcor-ip.net) |
| 01:53.08 | nzkiwi1 | I will be back in abt 10 minutes |
| 02:06.04 | nzkiwi1 | back |
| 02:07.28 | nzkiwi1 | googling tcptls.c:235 handle_tcptls_connection: FILE * open failed shows others have experienced it but has no useful info |
| 02:10.40 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.167.20.167) |
| 02:18.18 | dym | nzkiwi1: Then you're stuck with either continuing research, or posting on the asterisk forums/bugtrackers |
| 02:19.29 | ChrisInSydney | dym, nzkiwi1: Thats what I said |
| 02:19.37 | ChrisInSydney | :-/ still no luck then |
| 02:20.02 | nzkiwi1 | the blink client makes no difference |
| 02:20.23 | dym | ChrisInSydney: Nah, I had a look around, but no chance. |
| 02:20.39 | dym | nzkiwi1: Did you try to reduce the sip.conf to the very basics? |
| 02:20.51 | dym | you seem to have quite the bloated one |
| 02:21.36 | nzkiwi1 | my sip.conf is simple. you must have me confused |
| 02:21.46 | dym | nzkiwi1: well, you had tls settings on there |
| 02:21.50 | dym | which is by far no default |
| 02:22.42 | nzkiwi1 | the sip.conf is fine |
| 02:22.51 | dym | This is contraproductive. |
| 02:22.54 | dym | How can you be sure of that? |
| 02:23.12 | dym | Every forumpost regarding this issue was related to a false tls setting |
| 02:23.21 | dym | false or incompatible even |
| 02:23.31 | nzkiwi1 | which setting? |
| 02:23.57 | dym | Did you try breaking the config down to the very connection basics? |
| 02:25.24 | nzkiwi1 | I started with a tried and provne sip.conf and added in the settings as specified in the official documentation |
| 02:25.37 | nzkiwi1 | https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 02:25.43 | dym | provne? |
| 02:25.51 | nzkiwi1 | what? |
| 02:25.59 | dym | thats quoting you... |
| 02:26.11 | nzkiwi1 | provem |
| 02:26.14 | nzkiwi1 | proven |
| 02:26.19 | dym | ah |
| 02:26.29 | dym | proven as in - it worked for you without TLS? |
| 02:26.35 | nzkiwi1 | yes |
| 02:29.05 | nzkiwi1 | here are all the TLS settings in the [general] section: |
| 02:29.36 | nzkiwi1 | tlsenable = yes |
| 02:29.36 | nzkiwi1 | tlsbindaddr = 0.0.0.0 |
| 02:29.36 | nzkiwi1 | tlscertfile = /etc/asterisk/certs/server.pem |
| 02:29.37 | dym | STOP |
| 02:29.39 | dym | nopaste |
| 02:29.42 | dym | thanks |
| 02:29.47 | nzkiwi1 | tlscafile = /etc/asterisk/certs/ca.crt |
| 02:29.47 | nzkiwi1 | tlsdontverifyserver = yes |
| 02:29.47 | nzkiwi1 | tlsclientmethod = tlsv1 |
| 02:29.47 | nzkiwi1 | tlscipher = DES-CBC3-SHA |
| 02:29.50 | nzkiwi1 | that is it |
| 02:29.51 | dym | hey |
| 02:30.05 | dym | you dont just paste random content into the channel. use a nopaste next time. |
| 02:30.28 | nzkiwi1 | nopaste? |
| 02:30.37 | dym | yes, a website to paste code onto |
| 02:30.48 | dym | so you can post its link here and not flood the channel |
| 02:31.45 | nzkiwi1 | http://www.pastebin.com/HpgFdj9D |
| 02:31.48 | nzkiwi1 | sorry |
| 02:35.07 | dym | did you try commenting the tlscipher parT? |
| 02:36.04 | ChrisInSydney | nzkiwi1: I go slammed for pasting in something I didnt mean to. I cut two lines to past in, but F&^%ing oultlook decided I should have my previous paste form the clipboard or afond 20 lines of code |
| 02:36.22 | ChrisInSydney | embarrasment |
| 02:37.43 | nzkiwi1 | I just has a look at the file tcptls.c and by my redaing it was last updated in 2008 |
| 02:37.57 | nzkiwi1 | copyright 2007 - 2008 |
| 02:38.29 | nzkiwi1 | I have previously tried commenting the tlscipher part and no difference |
| 02:40.08 | nzkiwi1 | I have just tried that again and I am still getting 2 faults: File * open failed and the peer is unreachable. whether these two are elated or not I dont know |
| 02:40.17 | *** join/#asterisk afink (~afink@ip68-13-94-224.om.om.cox.net) |
| 02:40.22 | dym | please nopaste the output of "sip show peers" and "sip show registry" |
| 02:40.52 | dym | is your debug level set to 9? |
| 02:40.59 | dym | if not => core set debug 9 |
| 02:41.04 | nzkiwi1 | 0 right now, sometimes 60 |
| 02:42.01 | dym | no such level :D |
| 02:42.20 | nzkiwi1 | it says '0' sip registrations |
| 02:42.30 | dym | how is your phone connected? |
| 02:42.44 | dym | SIP? |
| 02:44.07 | nzkiwi1 | http://www.pastebin.com/UdGy4wX1 |
| 02:44.13 | nzkiwi1 | yes |
| 02:44.34 | dym | huh |
| 02:44.38 | dym | what is this output? |
| 02:44.58 | nzkiwi1 | which is weird because other phones are connected which work (they are set to UDP) |
| 02:45.30 | nzkiwi1 | that is the output of the testing peer |
| 02:45.49 | dym | i meant the actual command "sip show peers" |
| 02:48.04 | nzkiwi1 | http://www.pastebin.com/ZmEwkMyj |
| 02:48.14 | nzkiwi1 | that is a partial c&p |
| 02:48.24 | *** join/#asterisk afink (~afink@ip68-13-94-224.om.om.cox.net) |
| 02:48.24 | dijib | who wants a sangoma card. as a break from your tls woes |
| 02:48.35 | dym | which is the client in question? |
| 02:48.45 | nzkiwi1 | 199 |
| 02:48.45 | dym | dijib: which one? |
| 02:49.04 | nzkiwi1 | 199 is the test. 100 and 102 are set to UDP and work |
| 02:50.09 | dym | odd |
| 02:50.44 | nzkiwi1 | the pbx is multi homed. wan is public IP and lan is 192.168.0.0/24 |
| 02:50.52 | nzkiwi1 | no nat is used |
| 02:50.57 | dym | where is the client? |
| 02:51.05 | nzkiwi1 | 199 client? |
| 02:51.08 | dym | yes |
| 02:51.17 | nzkiwi1 | on a public IP |
| 02:51.32 | dym | so outside your net |
| 02:51.49 | nzkiwi1 | I have a range of public IPs |
| 02:51.52 | dijib | dym: a-200r |
| 02:52.55 | dym | age/price? |
| 02:56.37 | nzkiwi1 | has anyone here got tls working to a voip client or phone? could they please advise me which version of open ssl and asterisk they are using? |
| 02:57.03 | dym | nzkiwi1: want a good hint? |
| 02:57.11 | dym | post on the forums. |
| 02:57.55 | nzkiwi1 | been there done that. no worthwhile response |
| 02:58.28 | dym | you expect more from posting on IRC on the weekend? |
| 02:58.30 | dym | seriously? |
| 02:59.06 | nzkiwi1 | all the O.P's were on weekdays |
| 02:59.20 | dym | Can you link me to your post? |
| 03:00.15 | dijib | age, mfg 2007. it has a single FXO module so 2 of the 4 ports working, expandable upto 24ports |
| 03:00.33 | dijib | considering reasonable trade.and shipping |
| 03:00.45 | nzkiwi1 | it is presently at the top of asterisk support |
| 03:01.00 | dym | link? |
| 03:01.28 | dijib | link to what? i have no ad. i can surely find info on the card thouh if thats what you rlooking for |
| 03:01.37 | dijib | im looking myself for additional ata's |
| 03:01.43 | dym | dijib: chill. talking to nzkiwi1 |
| 03:02.14 | dijib | he needs to cut cheques. im here for the * social |
| 03:02.30 | dym | dijib: name your price |
| 03:02.44 | dijib | what services have you rendered? |
| 03:02.54 | dym | huh? |
| 03:02.55 | nzkiwi1 | http://forums.asterisk.org/viewtopic.php?f=1&t=81693 |
| 03:03.00 | dijib | to the aformentioned party |
| 03:03.34 | dym | that post is only 2 days old |
| 03:03.59 | Kobaz | hmm, if you ignore forwards in Dial (with the 'i' option), how would you get the forwarding destination in dialplan |
| 03:04.14 | dym | dijib: none whatsoever |
| 03:04.33 | Kobaz | there's a SIPLASTERRORNUM=302 and SIPLASTERRORTXT=Moved Temporarily |
| 03:04.35 | dym | nzkiwi1: i'd suggest to give it some time |
| 03:04.46 | Kobaz | and there doesn't seem to be a reference to where it moved to |
| 03:05.31 | nzkiwi1 | that is the most recent post. There was a couple before |
| 03:08.54 | dijib | typing takes times |
| 03:10.48 | dijib | i really wanna trade this card for something, i dont care if its a motor or whatever i could beinterested |
| 03:11.12 | nzkiwi1 | I only use VoIP here |
| 03:12.43 | dym | dijib: chocolatebar? |
| 03:12.56 | dym | Mars/Twix - your choice |
| 03:13.39 | Kobaz | hacks some asterisk |
| 03:23.03 | dijib | chocolattebar will not suffice, i already have one of those |
| 03:26.01 | dym | well |
| 03:26.09 | dym | i could eat half of it, pre-sending |
| 03:26.16 | dym | that'd make it pretty much unique |
| 03:26.29 | dym | also seen as probably no one else has the same teeth configuration |
| 03:26.34 | dym | how about now? |
| 03:41.52 | dijib | neither dym p3nguin would need more the deposition than you |
| 03:41.58 | dijib | appreciate |
| 03:42.02 | dijib | the |
| 03:42.31 | dym | whatnow? :D |
| 03:42.49 | dym | sit still, take a deeeeeeep breath, re-think what you were gonna say - THEN re-type :D |
| 03:42.52 | *** join/#asterisk ccesario_ (~ccesario@187.17.166.162) |
| 03:43.15 | dym | im gone now anyways :D |
| 03:43.43 | dijib | goodluck |
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| 04:41.17 | funkylonehat | Hi guys, anyone else having the chanspy == poor voice quality on asterisk 1.8? |
| 04:42.35 | funkylonehat | not much traffic? |
| 04:44.28 | *** join/#asterisk funkylonehat (7114566b@gateway/web/freenode/ip.113.20.86.107) |
| 04:44.42 | funkylonehat | Hi? |
| 04:56.40 | *** join/#asterisk funkylonehat (~funkylone@125-236-222-73.adsl.xtra.co.nz) |
| 04:56.48 | funkylonehat | Hi Leif |
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| 04:57.52 | funkylonehat | hi all, |
| 04:58.03 | funkylonehat | working on an issue with chanspy on 1.8 |
| 04:58.27 | funkylonehat | quality of the call deteriorates when using chanspy to monitor with just the b option |
| 04:58.41 | funkylonehat | never had issues with 1.4, recently migrated to 1.8.6 |
| 04:58.50 | funkylonehat | any pointers anyone? |
| 05:01.07 | funkylonehat | * timing = dahdi_dummy, transmit_silence=yes on asterisk.conf |
| 05:01.23 | funkylonehat | same results with and without monitor. on 1.4 same setup worked without issues |
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| 05:06.27 | funkylonehat | anyone here? |
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| 11:50.14 | Goni | I am facing hard time with this regex, can someone point out what I am doing wrong here? |
| 11:50.22 | Goni | exten => s,n,ExecIf($["${CALLERID(ani)}" = "${REGEX(^[a-zA-z0-9]+$)}"]?Set(CALLERID(ani)=1923)) |
| 11:50.45 | Goni | in my case if I have any a-z or A-Z in the CLI, I want to change it to something static |
| 11:51.33 | WIMPy | There go 6½ hours of silence. |
| 11:51.37 | Goni | :) |
| 11:52.19 | Goni | so you were actually reading out the silence :) |
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| 15:00.24 | dym | Oh, and here I thought there was activity on here |
| 15:00.33 | dym | Damn non-ideal highligh colours! |
| 15:05.15 | *** join/#asterisk azertyu (~chatzilla@28.23.74.86.rev.sfr.net) |
| 15:05.30 | azertyu | hello there |
| 15:05.57 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 15:06.47 | azertyu | i just followed this tutorial samyantoun.50webs.com/asterisk/freepbx/portechmv370/ |
| 15:07.07 | azertyu | after setup i can't able to make / receive calls |
| 15:07.41 | WIMPy | ~freepbx |
| 15:07.41 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:08.46 | azertyu | and here is the log |
| 15:08.57 | azertyu | asterisk cli log |
| 15:09.24 | azertyu | http://paste.ubuntu.com/848604/ |
| 15:09.52 | azertyu | i can't understand where is the problem ? can anyone help ? thanks |
| 15:10.00 | WIMPy | azertyu: You need to ask in #freepbx. They know how to configure that thing. (hopefully) |
| 15:10.44 | WIMPy | But looks like you tried to call an extension that doesn;t exist. |
| 15:11.15 | azertyu | what is the extension ? |
| 15:11.25 | azertyu | what extension ? |
| 15:12.04 | WIMPy | Have you tried to read the log you pasted yourself? |
| 15:13.12 | azertyu | yes of course |
| 15:13.20 | azertyu | about talking about 7777 extension? |
| 15:13.28 | WIMPy | yes |
| 15:13.59 | azertyu | have i need to create that in sip extension ? |
| 15:16.00 | WIMPy | extensions go to extensions.conf, but if you're using freepbx, you better not change that manually. |
| 15:18.25 | *** join/#asterisk neurosys_ (~neurosys@c-76-108-39-221.hsd1.fl.comcast.net) |
| 15:18.29 | dym | Good advice. |
| 15:19.08 | azertyu | WIMPy: how to with freepbx ? |
| 15:19.22 | dym | #freepbx <-- try asking there. |
| 15:19.24 | azertyu | & wt to do ? |
| 15:19.33 | dym | This is not a channel for freepbx advice. |
| 15:19.35 | azertyu | are you there ? |
| 15:19.43 | dym | azertyu: Ask on #freepbx |
| 15:19.58 | azertyu | no onz e there |
| 15:20.05 | dym | Then you'll have to wait. |
| 15:20.06 | azertyu | are you there ? |
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| 16:28.03 | mechbangirc | I am trying to slim down asterisk, Need only minimal functionality with realtime support. However I am using IF function and I could find out which module is required. I get an error IF is not registered. Asterisk version 1.6.2.22 |
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| 17:46.03 | Assid | hi |
| 17:46.33 | Assid | i have my android phone working fine.. for the most part.. but i seem to be facing an issue with using my nokia e71 with asterisk |
| 17:46.53 | Assid | if the call is incoming.. theres audio.. if its outgoing from the nokia.. theres NO audio whatsoever |
| 17:47.20 | Assid | i alos get the following: NOTICE[6017]: res_rtp_asterisk.c:2241 ast_rtp_read: Unknown RTP codec 127 received |
| 17:57.59 | Assid | anyone know any other sip client for nokia e71 |
| 18:04.03 | Assid | anyone know why im getting this error : Unknown RTP codec 127 received |
| 18:07.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 18:09.10 | Assid | sup [TK]D-Fender |
| 18:13.01 | Assid | anyone know why im getting this error : Unknown RTP codec 127 |
| 18:13.08 | Assid | its happening when im using my nokia e71 |
| 18:13.21 | Assid | is staring into thin air |
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| 18:26.47 | *** join/#asterisk veryhappy (~no@port-92-195-78-36.dynamic.qsc.de) |
| 18:27.22 | veryhappy | hi, got a question i installed also german on my asterisk a few files are still in english tho |
| 18:38.14 | [TK]D-Fender | veryhappy: yOU ARE SIMPLY MISSING CERTAIN FILES IN gERMAN AND IT IS FALLING BACK TO eNGLISH.. |
| 18:38.19 | [TK]D-Fender | Darn caps.... |
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| 18:44.50 | veryhappy | no problem [TK]D-Fender :D i also know the problem with the caps :D ok under which packet could i install this sounds? |
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| 18:47.13 | [TK]D-Fender | veryhappy: if you're missing a few.... well they're just missing.. it isn't another package...... |
| 18:47.31 | veryhappy | for example: vm-urgent |
| 18:47.42 | veryhappy | and the numbers on the voicemail |
| 18:48.03 | veryhappy | like "press 1 for new messages, 2 for folders...:" and so on |
| 18:48.23 | [TK]D-Fender | they are jsut missing. Go ask whoever packaged up those sounds |
| 18:48.40 | veryhappy | in the internet or where? |
| 18:48.42 | veryhappy | :D |
| 18:49.31 | [TK]D-Fender | "Go ask whoever packaged up those sounds" <----------- |
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| 19:23.31 | veryhappy | thx cya |
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| 20:30.13 | neurosys_ | anyone feel like helping what I would believe is a n00b issue. But i simply cant figure it out |
| 20:31.46 | neurosys_ | Have * behind a Cisco 1941. Turned ALG off and did port forwarding on ourt 5060 and 10000-20000. set externip to the ext ip, and localnet. Phones locally and remote register fine... |
| 20:32.23 | neurosys_ | But when remote tries to call thru the PBX, call connects but no audio. IF i call the PBX directly, audio works. But routing out of the trunk, no audio. |
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| 20:38.52 | ChrisInSydney | neurosys_: What port range have you got set in rtp.conf ? |
| 20:39.57 | neurosys_ | 10000-20000 |
| 20:40.09 | ChrisInSydney | nat=yes; canreinvite=no; directmedia=no ? |
| 20:40.37 | ChrisInSydney | and a SIP trunk ? |
| 20:40.51 | neurosys_ | <PROTECTED> |
| 20:41.19 | ChrisInSydney | its implied I assume ( someone will correct me if im wrong) |
| 20:42.02 | neurosys_ | Hmm im looking at the cisco router and I dont see the 10000-20000 range explicitly set to forward to the PBX. Think that could be the issue? |
| 20:42.17 | ChrisInSydney | sounds like you might me on to something |
| 20:42.27 | ChrisInSydney | :) |
| 20:42.51 | neurosys_ | There is an ACL to allow it, but not an actualy forward. |
| 20:43.32 | ChrisInSydney | ahh |
| 20:44.04 | ChrisInSydney | that'll do it |
| 20:44.50 | neurosys_ | So the reason it would work taking directly to the PBX is the RTP is generated by the PBX, but when connecting thru the trunk, the trunk initates RTP after the SIP ack? |
| 20:45.45 | ChrisInSydney | sounds plausable |
| 20:47.38 | ChrisInSydney | working now ?? |
| 20:47.55 | neurosys_ | Trying to remember how to fwd ranges in IOS |
| 20:48.02 | neurosys_ | :P |
| 20:49.16 | ChrisInSydney | I dont speak Cisco, I am learning though. I'm up to enable -> configure terminal. Then it all gets too confusing from there othe rthan that the tab key and the ? are my friends |
| 20:49.30 | ChrisInSydney | X/ |
| 20:49.59 | neurosys_ | hehe |
| 20:51.09 | ChrisInSydney | good luck. Have to go and get some "real work" done. Need a coffee first. 8am here |
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| 20:54.37 | neurosys_ | hmm turns out there is an explicit port forward. So im stuck again |
| 20:56.51 | ChrisInSydney | neurosys_: sip debug and look at the IP addressing SDP. Its hidden somewhere in there |
| 20:57.52 | neurosys_ | ChrisInSydney, searching... |
| 20:58.20 | neurosys_ | ChrisInSydney, looks correct to me |
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| 21:27.15 | helix9 | I've got a quicknet pci card, and I see it initialized in /var/log/dmesg, but I'm not getting any /dev/phone* devices |
| 21:27.31 | helix9 | Would it be a different name? |
| 21:27.50 | helix9 | It's an Internet Phonejack, also |
| 21:33.02 | ChrisInSydney | Thats why I usually run my extenal handsets through openVPN, or I have a system with a public IP and do some dialplan stuff there |
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| 21:35.11 | helix9 | was that for me Chris? |
| 21:35.15 | ChrisInSydney | neurosys_: Dont know what happened there. :-/ |
| 21:35.32 | ChrisInSydney | No for neurosys_ |
| 21:35.43 | ChrisInSydney | sorry |
| 21:36.01 | helix9 | no prob |
| 21:36.27 | ChrisInSydney | helix9: I probably cant help you with your issue. |
| 21:36.50 | ChrisInSydney | never seen / touch / smelt one of those. Are they a digium copy ? |
| 21:37.41 | helix9 | I don't think so, but I'm not sure |
| 21:37.56 | helix9 | it's a single port FXS pci card |
| 21:39.42 | ChrisInSydney | http://www.quicknet.net/ doesnt exisit anymore |
| 21:40.37 | helix9 | yeah, I bought the card off Ebay to see if I could give myself a free Google Voice landline |
| 21:40.57 | helix9 | I see it in lspci, and the ixj driver loads, but doesn't give me a /dev/phone device |
| 21:41.06 | ChrisInSydney | Im always inclinded to go SIP all the way and use an ATA |
| 21:41.14 | helix9 | an ATA? |
| 21:41.35 | ChrisInSydney | Analog(ue) telephone adapter |
| 21:41.45 | ChrisInSydney | SPA3102 for example |
| 21:42.51 | ChrisInSydney | http://www.asterisk.org/support/hardware says your card is supported. I also just read that you need at least a 486 66MHz CPU with a free PCI slot |
| 21:43.03 | ChrisInSydney | on another site |
| 21:43.36 | helix9 | yeah, it's a pretty old model |
| 21:43.44 | helix9 | from 2001, I think |
| 21:43.58 | ChrisInSydney | so am I, but I try not to let on too much |
| 21:44.17 | helix9 | lol |
| 21:44.24 | ChrisInSydney | I call it a mid life renaissance |
| 21:44.40 | ChrisInSydney | /dev/dsp |
| 21:45.06 | helix9 | I'm not seeing a /dev/dsp either |
| 21:46.24 | ChrisInSydney | The relevant configuration file is "phone.conf" and the channel name is |
| 21:46.25 | ChrisInSydney | Phone/phone0 |
| 21:46.33 | ChrisInSydney | exten = 999,1,Dial(Phone/phone0) |
| 21:46.43 | ChrisInSydney | just bots I am seeing |
| 21:46.45 | ChrisInSydney | bits |
| 21:46.57 | helix9 | thanks |
| 21:49.36 | ChrisInSydney | wow, Im looking at posts from 2001 |
| 21:50.45 | helix9 | yeah, I keep seeing articles where it said the device was changed from /dev/ixj to /dev/phone, but unfortunately I don't see either |
| 21:51.31 | ChrisInSydney | AFAIK, most of this stuff is wrapped up so that zaptel could talk to it. But zaptel has been replaced with dahdi. I am not too sure if the stuff like the quicknet and voicetronics cards works anymore |
| 21:52.34 | ChrisInSydney | if I was you i'd sell it and buy an unlocked SAP3102 which will give you pass through to your existing PSTN |
| 21:52.52 | ChrisInSydney | thats the wondeful thing about ebay |
| 21:53.00 | helix9 | yeah, I might have to do that |
| 21:53.30 | ChrisInSydney | you are pretty much guaranteed to get the amount of the guy that you beat back in your hand, unless you were the only bidder |
| 21:54.02 | helix9 | it was a buy it now auction |
| 21:54.12 | helix9 | hopefully I'll be able to sell it off again |
| 21:54.49 | ChrisInSydney | how much did it cost ? |
| 21:55.00 | helix9 | I paid about $60 for a lot of six cards |
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| 21:55.27 | helix9 | wish I had gone with an ATA instead |
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| 21:56.40 | ChrisInSydney | http://flylib.com/books/en/3.440.1.30/1/ |
| 21:56.48 | ChrisInSydney | has some stuff re zaptel |
| 21:57.05 | ChrisInSydney | You could try with zaptel and ast 1.4 |
| 21:57.42 | ChrisInSydney | must fly, good luck |
| 21:57.47 | Who-m3 | Looking for a bit of assistance. Trying to set up fax capabilities using Asterisk. Wanting to receive a fax on a set DID and it be translated into a tiff image, which can then be e-mailed as a pdf (I've got the conversion script ready). Every time I try to receive a fax, it fails. I've tried using FreeSwitch as a stand alone interface, as well as a bridge for my Google Voice number. |
| 21:57.47 | Who-m3 | Asterisk recognizes it's an attempted fax, but fails to move forward. I'm not sure what I've missed. |
| 21:58.04 | Who-m3 | Any assistance would be greatly appreciated, as I'm pulling my hair out now... |
| 21:58.28 | helix9 | thanks, I'll take a look |
| 21:59.10 | Who-m3 | Oh, running 1.8.9.2 on CentOS 5.7 64bit |
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| 22:22.57 | [TK]D-Fender | Who-m3: show us |
| 22:22.58 | [TK]D-Fender | ~pb |
| 22:22.59 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 22:23.03 | [TK]D-Fender | ^^^ |
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| 23:14.12 | jaytee | I knew I should have called the 800 number for Tractor-Trailer driving school instead of taking that first IT job 23 years ago. Damn you Microsoft and your crappy Exchange software! You're trying to kill me, I just know it! |
| 23:22.19 | afink_ | do you guys recommend using a different password for each internal extension in sip.conf? Or would one strong one suffice? |
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| 23:39.25 | ChrisInSydney | afink_: Depends. Are you on a public IP ? |
| 23:40.20 | afink_ | I am not but I have ports that pass through as though I am |
| 23:41.47 | ChrisInSydney | afink_: Have you got external handsets ?? |
| 23:41.55 | afink_ | no |
| 23:42.10 | afink_ | I have one over the internet SIP provider though |
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| 23:45.47 | ChrisInSydney | afink_: Are you likely to call in then redirect the call to a mobile / cell or another outside number ? |
| 23:46.18 | afink_ | yes |
| 23:46.42 | ChrisInSydney | I was hoping to cut the port forwarding for you. |
| 23:47.25 | afink_ | I will able to after I get rid of the one SIP provider and that will help immensely |
| 23:48.18 | afink_ | for now though, I have commented out all of the unused extensions, ACLed them, passworded and when I have time will change the username to not match the extensions |
| 23:50.32 | ChrisInSydney | I have used identical hard passwords > 14 characters and had no "real" problems |
| 23:56.42 | ChrisInSydney | afink_: Set up some router rules to limit the IP addresses that can connect on 5060 to your ITSPs subnets |
| 23:57.22 | afink_ | Great idea thanks |
| 23:58.55 | ChrisInSydney | Then you can relax a little bit more. Definately use ACLs and definately use different passwords to the SIP device names |