| 00:02.09 | p3nguin | jc319: Do they offer any sample configuration for Asterisk in their support section? I did a search and came up with no documents, but since I'm not signed in, there could be other things available to customers that I don't see. |
| 00:02.47 | jc319 | Yes, first I will find out if they are trying to do that really. If yes, end of contract. I will check voip.ms now. Is it one of your providers? |
| 00:03.37 | jc319 | There's an end user guide and an admin guide, I will check both in a minute but my guess is no, it was a pretty basic document last time I checked |
| 00:03.44 | p3nguin | Yes, I use VoIP.ms. |
| 00:04.16 | jc319 | I have the definitive list of UK numbers, so many templates... http://www.aa-asterisk.org.uk/index.php/Number_Format |
| 00:04.31 | p3nguin | I have at least three DIDs with VoIP.ms, and they've never once sent my calls to s. |
| 00:06.30 | jc319 | Perhaps I got it right about 8 hours ago, because I put _. and it was still not ringing. I never thought it could be an ITSP side issue. feeling a bit better now |
| 00:07.14 | p3nguin | _. is a poor choice for a pattern, but it DOES match 's' |
| 00:07.26 | p3nguin | It matches everything. That's why it's bad. |
| 00:08.05 | p3nguin | Go ahead and add the s extension like I mentioned above. At least that way you can get calls to one number. |
| 00:08.25 | p3nguin | It's better than no calls all weekend, I suppose. |
| 00:10.36 | jc319 | the problem is it did not work even with _. |
| 00:11.01 | p3nguin | After you add the s extension with a Goto(), see how it works out. |
| 00:11.13 | p3nguin | If you're still having problems, we can troubleshoot them. |
| 00:12.12 | jc319 | Anyway I will add 's' just to see. However first thing I would like to do is outbound calls, because noone knows this 3 DiD #s yet, it is my test system until it's tested and taken into production. So even if inbound does not work not a big problem, but I can start using outbound calls straight away, so that'll be good. |
| 00:12.18 | jc319 | I've added this >>> exten => _0[1-7]XXXXXXXXX,1,Dial(SIP/comscom_home/${EXTEN}); 11-digit dialing |
| 00:12.31 | jc319 | for example, a mobile number would be 07599401234 |
| 00:13.01 | jc319 | when I try calling it gave me this notice >> [Jun 11 00:09:02] NOTICE[3861]: chan_sip.c:21619 handle_request_invite: Call from '' to extension '07599401234' rejected because extension not found in context 'misc_calls'. |
| 00:13.53 | p3nguin | Your phone isn't matching the peer you configured for it. |
| 00:13.54 | jc319 | call from ''? Guess I need to fix my caller id too |
| 00:14.14 | jc319 | Why not? |
| 00:14.22 | p3nguin | Are you using a Cisco 7960 with a SIP firmware image? |
| 00:14.53 | jc319 | Yes |
| 00:15.06 | p3nguin | Are you serving config files for it from a tftpd? |
| 00:15.48 | jc319 | Yes |
| 00:16.34 | p3nguin | Do you have SIPDefault.cnf and SIP<MAC>.cnf in the tftp root? |
| 00:17.59 | jc319 | BTW, just tried with PC softphone, there's a different error msg there, a long one >> http://pastebin.com/sSzYCzt3 |
| 00:18.06 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 00:18.10 | jc319 | I think I do, checking now |
| 00:18.50 | carrar | SIPaRoooo |
| 00:20.05 | p3nguin | At first glance, that output looks like you call was successful as far as Asterisk, and then failed on the way out to the ITSP. |
| 00:22.07 | jc319 | Yes I have both, SIP<MAC>.cnf is what I edit all the time to change lines, I noticed I did not edit the default one for a few months, had a look now and changed media end port from 20624 to 20000. I'll factory reset the phones to activate changes |
| 00:23.05 | p3nguin | You can simply do a soft reset. |
| 00:23.57 | p3nguin | Just press and hold down *, 6, and Settings at the same time. |
| 00:24.00 | jc319 | Will do next time. While on the subject, when I change sip/ext.conf files; is 'sip reload' & 'dialplan reload' all I need? do I also need 'reload' or stop/start asterisk? |
| 00:24.25 | p3nguin | sip reload and dialplan reload are the appropriate commands. |
| 00:24.26 | jc319 | Yes I know that combination but I didn't know which settings stayed behind, so took extreme measures :D |
| 00:24.40 | jc319 | Great it'll take less time now. |
| 00:25.43 | p3nguin | It has been a while since I used a SIP image on my Cisco phones, so it may take me a minute to dig through my archive to remember what does what. |
| 00:27.13 | jc319 | media port fixed a typo about 2nd line (hangup) of numbering pattern fixed, and number pattern changed to _. and call still does not go through 7960 |
| 00:28.34 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 00:28.48 | p3nguin | I have no clue what you're trying to accomplish by using the _. pattern. |
| 00:30.13 | p3nguin | If you're still trying to get a call TO the phone, make sure the phone is registered by using "sip show peers" on the CLI. |
| 00:30.46 | p3nguin | If the phone does not have an IP address listed, Asterisk has no idea how to reach it. |
| 00:31.01 | jc319 | just testing with PC softphone, it does not go through either. http://pastebin.com/Spth8zdm (same 32000 ms timeout msg) |
| 00:32.05 | p3nguin | SIP/comscom_home-0000001d is circuit-busy |
| 00:32.19 | jc319 | My first goal is to being able to make calls on hardphones and softphones. At the moment nothing works |
| 00:32.20 | p3nguin | They are not accepting the call right now. |
| 00:32.38 | p3nguin | Add an echo test to verify the phone can call Asterisk. |
| 00:32.45 | jc319 | So they deliver calls to 's' and don't accept outbound calls? awesome |
| 00:33.01 | jc319 | Ok |
| 00:34.08 | p3nguin | http://pastebin.com/gwsd38xV |
| 00:34.50 | p3nguin | I'd drop that into the phones context or something so all your phones have access to it. |
| 00:39.00 | jc319 | After factory defaults reset, 7960's line icon is displayin 'disconnected' status |
| 00:39.28 | jc319 | in 'sip show peers' only the pc softphone has IP |
| 00:40.16 | p3nguin | Then that's the only phone that Asterisk knows how to reach. |
| 00:41.23 | jc319 | Is SNTP mandatory? I didn't install it (yet). However the line icon was connected before, so it shouldn't be the reason.. |
| 00:41.48 | p3nguin | Would you mind pasting your SIPDefault.cnf and SIP<MAC>.cnf files? Mask the passwords if necessary. |
| 00:42.44 | jc319 | Oups, got it, sorry! This phone on my desk has DiD #3 on it. I am testing with DiD #1 now. (I've removed 2nd and 3rd DiDs and additional (internal) sip peers while I was editing your configs, just to simplify). |
| 00:43.10 | jc319 | I will change this phone to the correct line and try again. |
| 00:43.58 | p3nguin | The phone should have no concept of a DID. |
| 00:44.44 | p3nguin | The phone just has the peer info to talk to Asterisk. Asterisk handles the calls to the DID number and sends them to the phones as instructed by dial plan. |
| 00:45.19 | jc319 | OK you're right, I managed to get confused by the label on the line. I better paste it :) thx for having a look in 5 mins |
| 00:48.51 | jc319 | SIP<MAC>.cnf @ http://pastebin.com/g1Rj5TDN SIPDefault.cnf @ http://pastebin.com/10ZXf34f |
| 00:51.45 | jc319 | and the latest copies of .confs are here -- sip.conf @ http://pastebin.com/KjZPrVY0 extensions.conf @ http://pastebin.com/zkURQFzM |
| 00:52.40 | jc319 | At the moment PC softphone can enter echo test, 7960 cannot ([Jun 11 00:52:15] NOTICE[3861]: chan_sip.c:21619 handle_request_invite: Call from '' to extension '500' rejected because extension not found in context 'misc_calls'.) |
| 00:53.20 | p3nguin | You're going to need to add insecure=port into the peer entries for the Cisco phones. |
| 00:55.01 | jc319 | Done. Reloaded, pc softphone still has a host (added that insecure=port to that one as well), 7960s still do not have a host |
| 00:55.22 | p3nguin | Give them a minute to register. |
| 00:55.31 | jc319 | shall I soft reboot |
| 00:55.48 | p3nguin | I suppose you could... or you can just wait a moment. |
| 01:00.02 | jc319 | Rebooted one but still no IP in sip show peers screen. |
| 01:00.14 | jc319 | [Jun 11 00:57:24] WARNING[3861]: chan_sip.c:3580 retrans_pkt: Hanging up call 7d10bd2153baa76055f15125242c8b9c@78.86.169.203:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). |
| 01:00.20 | jc319 | <PROTECTED> |
| 01:00.20 | jc319 | <PROTECTED> |
| 01:00.20 | jc319 | <PROTECTED> |
| 01:00.20 | jc319 | <PROTECTED> |
| 01:01.13 | jc319 | Also this is still the same... |
| 01:01.19 | p3nguin | The only time I can think of seeing the "no reply to our critical packet" warning is when there is a NAT issue. |
| 01:03.16 | jc319 | On this side of the NAT, PC softphone can call 500 echo test, 7960s still cannot. (still no IP in show peers) |
| 01:03.47 | jc319 | so this means 7960s do not register to Asterisk, right? Why would they not register |
| 01:04.14 | p3nguin | Do you have a NAT between Asterisk and the phones? I noticed more nat=yes settings where I don't think they need to be. |
| 01:05.14 | p3nguin | If you want to see if they are even trying to register, you'll need to perform a sip debug operation. "sip set debug on" should turn sip debugging on for you. |
| 01:06.33 | jc319 | It says Unauthorized http://pastebin.com/d40bYgr0 |
| 01:08.57 | jc319 | I changed all nat='s to NO and now 7960s can reach 500 |
| 01:10.35 | p3nguin | Here's my sip definition from back when I used SIP on my Cisco phone: http://pastebin.com/JbEa4jsv |
| 01:10.51 | p3nguin | Use it for reference. |
| 01:21.18 | p3nguin | This won't make any difference on the issues you've been having... |
| 01:21.38 | jc319 | Modified and using as template now. |
| 01:21.43 | p3nguin | SIP 8.12 has been reported to have some kind of caller ID bug. |
| 01:22.21 | p3nguin | If you have 8.11 available, you might consider using it once you straighten out the other configuration problems. |
| 01:26.09 | jc319 | Noted thanks. I would like to try the 's' modification now to see if I can get inbound calls. Just adding a new extension with the name 's', is that it? |
| 01:26.39 | jc319 | And in which context should I add it? |
| 01:27.08 | jc319 | (or just edit 'default' to ring a particular sip peer ?) |
| 01:28.09 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:29.10 | *** join/#asterisk azam (~azam@h-91-120.A163.priv.bahnhof.se) |
| 01:30.45 | p3nguin | Add the s extension to the inbound context and make it Goto() the other phone number. There is no "default" unless you created an extension called "default". |
| 01:32.12 | p3nguin | exten => s,1,Goto(other-number,1); where other-number is your actual phone number that you already created an extension for. |
| 01:32.12 | azam | hi, i am using g729 codec in pass through mode but when the called party indicates ringing, the calling party does not hear that. and i get a warning in asterisk logs. " channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)" |
| 01:32.41 | azam | my question: is there any sound file which plays ringing sound? if yes then i can convert that file to g729 also and solve my issue. |
| 01:32.56 | p3nguin | Ringing() |
| 01:33.16 | p3nguin | It's not a sound file, but it is ringing. |
| 01:33.38 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 01:34.42 | azam | p3nguin: sorry i dont understand. as far as i knw Ringing is an app to play ringing sound. it wont work in my senario. |
| 01:34.57 | p3nguin | It's an app, yes. |
| 01:35.32 | azam | but u see i have already dialed out the number. and ringing notification is sent to my asterisk from termination server. |
| 01:35.54 | p3nguin | yeah |
| 01:37.01 | azam | i saw that there is R option in dial but that is available only if you are using kapejod's Bristuff. |
| 01:37.31 | p3nguin | The r option is also available, but it's not going to be much different from Ringing(). |
| 01:37.58 | azam | p3nguin: sorry i am not an expert on asterisk. can u please elaborate a little on how i can use ringing app? |
| 01:38.09 | azam | let me tell u the whole senario. |
| 01:38.45 | p3nguin | exten => 0,1,Ringing(); |
| 01:39.02 | p3nguin | exten => 0,n,Wait(4); |
| 01:39.15 | azam | we get a call from a provider. i call my own application through dial plan, then from with in that application i dialout using dial app. |
| 01:39.29 | p3nguin | That would allow the Ringing app to run for 4 seconds on extension 0. |
| 01:41.13 | azam | that wud not be wise to play ringing just like tht bcuz then if the called number is busy, the calling user will hear something like RING RING BEEP BEEP BEEP |
| 01:41.22 | p3nguin | yep |
| 01:41.33 | azam | so :) |
| 01:41.35 | p3nguin | That's why we don't play ringing sounds unless we know what we're doing. |
| 01:42.12 | azam | yes i want to play ringing sound when i get ringing sip notification from termination carrier. |
| 01:42.51 | p3nguin | Do you have an aversion to the r Dial option? |
| 01:43.09 | azam | i knw if i have a codec for g729 then asterisk somehow plays the ringing tone to calling party when asterisk gets the notification. |
| 01:44.09 | azam | yes bcuz i dont want to play a ringing tone until i get one. |
| 01:44.27 | azam | that is simply unprofessional and confusing for the user to hear different tones. |
| 01:44.46 | p3nguin | I don't know how you can have it both ways. |
| 01:45.10 | azam | both ways? |
| 01:45.10 | p3nguin | You don't want to play a ringing sound because it is confusing and unprofessional |
| 01:45.12 | p3nguin | |
| 01:45.35 | p3nguin | You don't want to play a ringing sound because it is confusing and unprofessional, but you want to play ringing sounds to the caller. You can't have it both ways. |
| 01:46.05 | azam | sorry i think i didnt make myself clear. i want to play the ringing sound. but i only want to play it when the destination number is actually ringing. not before that. |
| 01:46.58 | azam | if i use Ringing() or 'r' then the ringing tone will be played as soon as the number is dialed out. |
| 01:47.13 | p3nguin | Like I said, you can't have it both ways. |
| 02:15.42 | *** join/#asterisk jeffgus (~jeffgus@green.zimage.com) |
| 02:30.04 | jc319 | p3nguin: 's' still does not work. I just checked the dialplan, there is another 's' at the top, it comes from app_dial. Is it possible & safe to remove it? http://pastebin.com/7UHQD2Wp |
| 02:31.46 | p3nguin | I've never even heard of context app_dial_gosub_virtual_context. |
| 02:34.15 | jc319 | there's a bug involving this but doesn't help much in my case https://bugs.digium.com/view.php?id=19301&nbn=4 |
| 02:35.29 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 02:39.58 | jc319 | [ Context 'comscom_home-inbound' created by 'pbx_config' ] 's' => 1. Dial(SIP/213&SIP/000AB747A418&SIP/001A6CA3696C,30) [pbx_config] |
| 02:40.23 | jc319 | is this not supposed to catch all inbounds in this context and ring the 3 SIP peers? |
| 02:41.04 | p3nguin | It's extension 's' ... not a catch-all. |
| 02:43.15 | jc319 | Right. In my case they send 's' so it is supposed to catch it, right? It does not... I disabled that app_dial.so module, so I thought maybe that's why it lost Dial() capability. However it cannot even do this>> exten => s,1,Playback(demo-echotest); |
| 02:43.38 | jc319 | I think this PBX doesn't work with this ITSP at least with current settings |
| 02:44.26 | jc319 | Is there any few minutes free / trial sip providers? just to test if it is my Asterisk installation that is broken or is it my ITSP? |
| 02:44.44 | p3nguin | Turn on sip debugging and make a call to the phone number. Paste the output for me. |
| 02:46.36 | p3nguin | You can have as many 's' extensions as you want, as long as they are in different contexts. When a call matches a peer definition in sip.conf, that call has to obey the context value (such as context=voipms-inbound) and that's where the call goes. |
| 02:46.58 | jc319 | Yes just noticed that and put the modules back in (perhaps they're doing something useful...). |
| 02:47.32 | p3nguin | So how about that sip debug? |
| 02:47.49 | p3nguin | The sip debug will show the process of the call when it comes in. |
| 02:47.59 | jc319 | Yes testing now as we speak |
| 02:48.08 | p3nguin | It will show if a peer is matched, what extension the call is going to, and in what context it has landed. |
| 02:49.19 | jc319 | OK debug is on. It is in 37 verbosity level. Called from mobile to DiD. (DiD show as registered in sip show registry) Nothing moved on Asterisk screen. |
| 02:49.36 | p3nguin | 37? That's kind of silly. |
| 02:50.05 | jc319 | It's 2 seconds press-n-hold |
| 02:50.19 | p3nguin | huh? |
| 02:50.23 | jc319 | Verbosity is at least 77 |
| 02:50.34 | jc319 | asterisk -rvvvv[holding finger on V key here] |
| 02:50.44 | p3nguin | Verbosity does not increase above 4, which means 4 will have the same output 37 which will have the same verbosity as 77. |
| 02:51.09 | p3nguin | Run asterisk as a daemon and let it be. |
| 02:51.40 | p3nguin | Connect to the CLI and set your verbose level as needed. |
| 02:52.04 | p3nguin | Reload what needs reloaded when it needs reloaded. |
| 02:53.16 | jc319 | OK so, 'core stop now' and then 'asterisk' only. then 'asterisk -r'. then 'core set verbose 2' |
| 02:53.18 | jc319 | Verbosity was 0 and is now 2 |
| 02:53.35 | p3nguin | Anyway, the sip debug should show a call if a call is coming in. You should see the message about looking for 's' in some context like you saw before. |
| 02:54.36 | p3nguin | I have no idea what OS you're using, but "asterisk" isn't usually the best way to run the daemon. There's usually an init script that does a better job. /etc/init.d/asterisk start |
| 02:55.40 | jc319 | It is CentOS running on HP Thin Client t5710. I'll look for the script was not aware of its existence. |
| 02:57.55 | p3nguin | My rc script runs asterisk as "/usr/sbin/asterisk -G asterisk -U asterisk" |
| 02:58.23 | p3nguin | If I would run it as "asterisk", it would probably be running as root instead of the proper user/group. |
| 02:58.51 | p3nguin | But that's not the cause of the lack of call processing. |
| 03:01.31 | jc319 | Testing 3rd post from bottom here http://forums.asterisk.org/viewtopic.php?f=1&t=73249&start=0 |
| 03:01.55 | jc319 | Yes the call does not even land on Asterisk. I am starting to think it is the evil ITSP. |
| 03:02.28 | p3nguin | You've verified it doesn't reach Asterisk by the sip debug? |
| 03:02.34 | jc319 | I can't remember clearly but I guess it was landing before tonight, was it not? |
| 03:02.47 | jc319 | Do you remember anything about it? [yes I did verify twice] |
| 03:03.04 | p3nguin | It was reaching Asterisk, going to the s extension, within the correct context. |
| 03:03.12 | jc319 | Nothing moves on screen when I have 'the person you've called is currently unreachable' on my mobile |
| 03:03.17 | p3nguin | I have a theory. What is your host setting for the peer? |
| 03:03.25 | jc319 | 1 sec |
| 03:03.30 | p3nguin | I don't have your old sip.conf paste. |
| 03:03.53 | jc319 | host=proxy.coms.com |
| 03:04.11 | p3nguin | proxy.coms.com has address 85.90.225.100 |
| 03:04.16 | p3nguin | Okay, so that's not it. |
| 03:04.34 | p3nguin | I was thinking it probably had more the one IP address, but you only provide one. |
| 03:05.00 | p3nguin | Can you also show me your ACL lines? That's the deny/permit pair. |
| 03:05.02 | jc319 | Note - If your DNS does not support the use of SRV records then you should instead use the host name sip.coms.com. If your VoIP phone is unable to resolve any DNS data due to local networking problems then you should use the IP address 85.90.225.100. |
| 03:05.15 | jc319 | This is what they sent before, so it is correct IP then... :( |
| 03:05.25 | jc319 | Yes sure |
| 03:05.42 | p3nguin | proxy.coms.com and sip.coms.com are both 85.90.225.100 |
| 03:05.51 | jc319 | deny=0.0.0.0/0.0.0.0 |
| 03:05.53 | jc319 | permit=85.90.225.0/24 |
| 03:05.53 | jc319 | permit=91.220.180.0/24 |
| 03:06.14 | p3nguin | What's the purpose of the second one? |
| 03:06.16 | jc319 | phones have this: |
| 03:06.18 | jc319 | deny=0.0.0.0/0.0.0.0 |
| 03:06.18 | jc319 | permit=10.9.8.0/255.255.255.0 |
| 03:06.45 | jc319 | This is from their email: |
| 03:06.46 | jc319 | For best results set your firewall to accept all inbound traffic on all ports from the Coms IP address ranges 85.90.225.0 to 85.90.225.127 and 91.220.180.0 to 91.220.180.255. |
| 03:07.25 | p3nguin | You can safely remove permit=91.220.180.0/24 until the address of proxy.coms.com and sip.coms.com change. |
| 03:08.14 | jc319 | Probably it's their disaster recovery site or something like that, I can keep for the time being (not to worry about it later should it go down while I'm on holiday etc) |
| 03:08.16 | jc319 | 2080 ? S 0:00 /bin/sh /usr/sbin/safe_asterisk |
| 03:08.17 | jc319 | 2094 ? Sl 0:04 /usr/sbin/asterisk -f -vvvg -c |
| 03:08.17 | p3nguin | This is, of course, more housekeeping. That value will not prevent you from getting a call on the correct IP address. |
| 03:08.18 | jc319 | ok that script works |
| 03:10.02 | p3nguin | I don't know if you've tinkered in the firewall lately or not, but double check that you have UDP 5060 and the UDP range as configured in rtp.conf (normally UDP 10000-20000) all still forwarded to Asterisk's IP address. |
| 03:14.17 | jc319 | It's the default 10k-20k range in rtp.conf and I checked the firewall 2 hours ago but will have another look now |
| 03:15.10 | jc319 | Yes fw is ok.. |
| 03:15.55 | p3nguin | Can you show me your current sip.conf? |
| 03:19.22 | jc319 | Sure, sorry was busy adding dhcpd,named to autostart |
| 03:19.27 | jc319 | pasting 1 sec |
| 03:20.07 | p3nguin | You're going to run those heavy services on your Asterisk system? |
| 03:20.51 | jc319 | http://pastebin.com/FesXWbHY |
| 03:21.46 | p3nguin | externaddr still has a port number. |
| 03:22.09 | jc319 | Well, for the foreseeable future there will be 2 users and very limited load on this Asterisk system so I think it would be OK. If not, I can move them to another terminal client or in worst case I can build a fanless server. I think performance won't be an issue [but I will keep an eye on it]. |
| 03:22.13 | p3nguin | directmedia= is present but does not have a value. |
| 03:22.37 | jc319 | Yes externaddr has a port but I checked the sample .conf document, it should be fine |
| 03:22.39 | p3nguin | Are you trying to create a gateway device out of that computer? |
| 03:24.43 | jc319 | directmedia, from what I read here http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite does not matter in my case |
| 03:27.14 | p3nguin | I don't see anything that should prevent the call from showing up. |
| 03:27.55 | jc319 | Ok restarted the thin server, dhcpd/named/asterisk (with safe script) all auto-start after boot. sip set debug on > called in > nothing happens |
| 03:28.38 | p3nguin | Are you trying to create a gateway device out of that computer? |
| 03:29.08 | jc319 | Gateway between what? I just want to make phone calls :) |
| 03:29.11 | p3nguin | There's no other reason you would need dhcpd and named running there, right? |
| 03:29.52 | p3nguin | I can't think of any other reason. |
| 03:30.04 | jc319 | IP phones receive ip address details & tftpt from this thin linux box |
| 03:30.09 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 03:30.21 | p3nguin | You don't already have a dhcpd on the LAN? |
| 03:30.23 | jc319 | Some new messages appeared: http://pastebin.com/yfed4Vsx |
| 03:31.20 | p3nguin | Okay, so it registered to the ITSP. Your call might arrive now. |
| 03:31.22 | jc319 | I had the one on router but turned it off, so that I can control the IPs phones got (I just like to see them at static IPs) easier to distinguish which device is doing whateveritis that appears in the logs |
| 03:31.48 | p3nguin | and named? |
| 03:31.49 | jc319 | I keep trying, but no it does not.. |
| 03:33.06 | jc319 | root 2037 0.0 0.4 2860 1024 ? Ss 03:21 0:00 /usr/sbin/dhcpd |
| 03:33.08 | jc319 | named 1743 0.2 1.4 39156 3468 ? Ssl 03:21 0:01 /usr/sbin/named -u named |
| 03:33.12 | p3nguin | ... |
| 03:33.14 | jc319 | it is not that heavy by the way |
| 03:33.21 | p3nguin | Yeah, but that's not the reason I'm asking. |
| 03:36.51 | jc319 | Why is it then? Completed that 'directmedia=' to nonat, still the same |
| 03:43.22 | jc319 | On ITSP web portal, all my calls from mobile to (supposedly asterisk) are listed as voicemail |
| 03:45.15 | linuxgecko | http://pastebin.com/asFYPRUh this is enough of a dialplan if, all i want to do is make google voice calls, yes? |
| 03:45.49 | linuxgecko | that's the entire contents of my extensions.conf |
| 04:00.55 | *** join/#asterisk jeffgus (~jeffgus@green.zimage.com) |
| 04:07.13 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 04:13.36 | jc319 | p3nguin: Thanks for assistance, leaving now, I'll call them on Monday looks like it's on their side... |
| 04:35.47 | Katty | hi |
| 04:36.01 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 04:55.36 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.181.113.104) |
| 05:00.31 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 05:08.34 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 05:14.19 | *** join/#asterisk echelon (~echelon@gateway/tor-sasl/harel) |
| 05:15.01 | echelon | hi |
| 05:15.06 | echelon | anyone use verizon wireless? |
| 05:15.57 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 05:21.39 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 05:26.35 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 05:34.52 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 05:41.59 | *** join/#asterisk golikwid|mac_ (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 05:48.35 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 06:07.40 | *** join/#asterisk p_masho (~mash@84.45.225.28) |
| 06:07.43 | *** join/#asterisk irroot (~irroot@197.107.213.33) |
| 06:08.08 | *** join/#asterisk Defraz (~Defraz@72-24-19-221.cpe.cableone.net) |
| 06:09.13 | irroot | is not here |
| 06:09.40 | *** part/#asterisk echelon (~echelon@gateway/tor-sasl/harel) |
| 06:09.55 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 06:13.17 | *** join/#asterisk Defraz (~Defraz@72-24-19-221.cpe.cableone.net) |
| 06:22.02 | p3nguin | Thanks for letting us know. Now I can sleep better. |
| 06:24.06 | irroot | lol p3nguin more trying to convince myself |
| 06:24.14 | irroot | its 8:30 |
| 06:24.26 | p3nguin | Your clock is off. |
| 06:24.53 | irroot | winters morning are not too be messed with ... |
| 06:27.04 | *** join/#asterisk Defraz (~Defraz@65.39.108.7) |
| 06:32.38 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 06:42.45 | *** join/#asterisk golikwid|mac_ (~chrislees@207.30.30.130) |
| 06:43.19 | *** join/#asterisk southskies_ (~southskie@ip70-190-172-186.ph.ph.cox.net) |
| 06:45.17 | *** join/#asterisk southskies (~southskie@ip70-190-172-186.ph.ph.cox.net) |
| 06:59.28 | southskies | Howdy! |
| 07:00.13 | southskies | I am seeing sudden unidirectional audio for all IAX2 channels after a few hours of operation. Thoughts? |
| 07:00.45 | southskies | OS is gentoo, load is 10% per proc. |
| 07:00.49 | southskies | x64 |
| 07:01.17 | *** join/#asterisk golikwid|mac (~chrislees@207.30.30.130) |
| 07:03.13 | southskies | howdy |
| 07:19.54 | *** join/#asterisk golikwid|mac (~chrislees@207.30.30.130) |
| 07:20.01 | *** join/#asterisk Defraz (~Defraz@65.39.108.7) |
| 08:31.27 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 08:32.21 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc7-pool12-2-0-cust219.15-1.cable.virginmedia.com) |
| 09:01.21 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 09:02.17 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 09:23.41 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 09:36.22 | *** join/#asterisk hetii (~hetii@87.99.51.172) |
| 09:36.27 | hetii | Hello :) |
| 09:39.01 | *** join/#asterisk irroot (~irroot@197.107.213.33) |
| 09:40.28 | hetii | irroot: :) |
| 09:41.02 | irroot | hey how oes |
| 09:41.04 | irroot | goes |
| 09:41.05 | hetii | we have next perfect day to play with t38modem :) |
| 09:41.41 | irroot | lol |
| 09:41.45 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 09:41.57 | hetii | i change a little route string and asterisk is able to recive call :) |
| 09:42.36 | irroot | cool thought it be something like that |
| 09:42.41 | irroot | nice |
| 09:43.19 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 09:43.33 | hetii | yep :) after my hard work i will wrote some nice well documented howto, it is madness that i spend so much time to run this stuff :) |
| 09:44.38 | irroot | when you done ill get you to play with T38 GW ... :D |
| 10:00.58 | *** join/#asterisk carloimperia (~carloimpe@93.56.144.115) |
| 10:07.11 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 10:20.47 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 10:23.37 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 10:26.45 | hetii | irroot: t38modem help show this syntax: --sip-register [user@]registrar[,pwd[,contact[,realm[,authID]]]] |
| 10:27.09 | hetii | what is contact,realm, and authid ? |
| 10:28.05 | hetii | i never ever can understart for what complicate so much easy things. when somone provide user and pass it should be enought for authorization process... :/ |
| 10:28.19 | irroot | you should be able to ignore them |
| 10:28.38 | WIMPy | Uh, still on to that one? |
| 10:28.44 | WIMPy | Good morning everyone! |
| 10:29.06 | irroot | morning |
| 10:29.10 | hetii | Morning :) |
| 10:29.21 | hetii | unfortunatly i got REGISTER transaction id=z9hG4bK94ea9426-8392-e011-8d9e-000e2e472643 timeout, making retry 5, timeout 4.000 |
| 10:29.24 | irroot | should only need un/pw/registrar |
| 10:29.40 | hetii | i try different arguments but without success |
| 10:30.10 | irroot | ok that is not getting a reply |
| 10:30.15 | irroot | check ports / ip ?? |
| 10:30.26 | irroot | its going out but not coming bac |
| 10:31.58 | hetii | hmm you know when i disable secret argument for this extension on * i can make call , on other hand its a loopback interface so there should be no problem with trafic |
| 10:35.42 | irroot | it may be for incoming calls only ?? |
| 10:36.06 | irroot | register with * to recieve calls |
| 10:36.13 | hetii | hmm |
| 10:36.19 | irroot | making calls is possible without registration |
| 10:36.42 | hetii | not so much. |
| 10:37.00 | WIMPy | Yes, registration is only to tell Asterisk where to send calls if you've got host=dynamic. |
| 10:37.29 | hetii | i set host=127.0.0.1 |
| 10:37.35 | hetii | in * |
| 10:37.40 | WIMPy | Then you can't register. |
| 10:37.53 | hetii | why? |
| 10:38.07 | WIMPy | It's not dynamic. |
| 10:38.22 | WIMPy | But Asterisk should tell you so. |
| 10:38.44 | hetii | but first of all we need to clear few points. t38modem listen on 127.0.0.1:6060 asterisk on 127.0.0.1:5060 |
| 10:39.06 | hetii | and now i do call from t38modem to * |
| 10:39.21 | WIMPy | Just configure that (I guess you already have) and forget about registering. |
| 10:39.29 | hetii | in * i set password for extension 100600 |
| 10:39.49 | hetii | so somehow somewher in t38modem configuration i should provide this password |
| 10:40.07 | WIMPy | Yes |
| 10:40.26 | WIMPy | But that's not related to registering. |
| 10:40.31 | hetii | yep the question is where |
| 10:40.53 | WIMPy | can't help you on that one. |
| 10:41.18 | hetii | for sip options i have: --sip-proxy [user:[pwd]@]host and --sip-register [user@]registrar[,pwd[,contact[,realm[,authID]]]] |
| 10:41.19 | WIMPy | But you can restrict the peer to 127.0.0.1 and not use user/password. |
| 10:41.50 | WIMPy | Ok, so that should be the proxy option then for un/pw. |
| 10:41.56 | irroot | set the --sip-proxy bits ?? |
| 10:43.01 | hetii | ok and the host= argument set to dynamic in * ? |
| 10:43.11 | WIMPy | no |
| 10:43.31 | hetii | so waht host0127.0.0.1 ? |
| 10:43.42 | hetii | *host=127.0.0.1 |
| 10:43.46 | WIMPy | sure |
| 10:43.56 | WIMPy | And don't forget the port. |
| 10:44.16 | hetii | for ? |
| 10:44.45 | WIMPy | The peers port. Otherwise * will use 5060 and talk to itself. |
| 10:45.20 | hetii | ah yes yes the port is set in * to 6060 |
| 10:45.31 | WIMPy | good |
| 10:46.03 | irroot | heti make sure you put a listen option on t38modem to make the port 6060 only |
| 10:50.28 | hetii | this is my configuration: http://pastebin.com/c3DsMm8U |
| 10:50.36 | hetii | and i got: 2011/06/11 12:50:09.927 Housekeeper:0xd697f700SIPINVITE transaction id=z9hG4bKbac1903e-8692-e011-88c6-000e2e472643 timeout, making retry 1, timeout 1.000 |
| 10:52.31 | WIMPy | Exactely like that? |
| 10:52.51 | WIMPy | Those * surely need escaping you quoting. |
| 10:53.28 | WIMPy | What does Asterisk tell when t38modem sends the INVITE? |
| 10:53.51 | WIMPy | s/you/or/ |
| 10:56.12 | hetii | http://pastebin.com/6tn6Rxf7 |
| 10:56.43 | hetii | but im not sure if its when invite is send or general |
| 10:56.49 | WIMPy | I don't see any INVITE there. |
| 10:57.40 | hetii | yep me to:) |
| 10:57.46 | hetii | wait i will change debug level |
| 10:59.27 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 11:00.26 | hetii | all what i see: http://pastebin.com/hFdNis9H |
| 11:01.25 | WIMPy | Still no call attempt in either log. |
| 11:08.29 | hetii | it was just after restarting t38modem and now full with call attempt |
| 11:08.29 | hetii | http://pastebin.com/C4npFH0h |
| 11:08.29 | hetii | on * i dont see activity when i try make call |
| 11:12.43 | *** join/#asterisk vfabi (~fabi@host-static-188-237-240-50.moldtelecom.md) |
| 11:13.42 | WIMPy | t38modem seems to change it's port to 56402. |
| 11:15.12 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 11:15.16 | hetii | hmm |
| 11:15.42 | hetii | <PROTECTED> |
| 11:15.44 | hetii | udp 0 0 127.0.0.1:6060 0.0.0.0:* 25102/t38modem |
| 11:16.08 | WIMPy | It probably generates a dynamic port per call. |
| 11:18.29 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 11:18.49 | hetii | Sending PDU (807 bytes) to: rem=udp$127.0.0.1:5060,local=udp$127.0.0.1:50524,if=127.0.0.1%lo |
| 11:19.03 | hetii | so remote it is ok |
| 11:19.50 | *** join/#asterisk Denial (~Denial@drgi.co.uk) |
| 11:20.13 | WIMPy | Maybe that's a case for insecure=port. But someone better at sip config should tell you. |
| 11:20.59 | irroot | ok |
| 11:21.28 | irroot | so it sends the invite out to asterisk ok |
| 11:21.41 | irroot | that last paste is the output from t38modem |
| 11:21.48 | irroot | but asterisk never gets it |
| 11:21.56 | WIMPy | Or maybe you can tell t38modem to only use one SIP port. |
| 11:22.48 | hetii | but you always have a local and remote port it is not bug |
| 11:23.27 | hetii | so the question is why asterisk don`t get this INVITE |
| 11:23.33 | sezuan | Is it possible to JabberSend a message from a certain jid? I've configured a transport foo.localhost and want to send a message from nick@foo.localhost This seems possible with aji_send_raw_chat but I haven't found a proper way to access this function without modifying the source code. |
| 11:23.37 | irroot | hetii a wireshark/tcpdump of port 5060 ?? |
| 11:23.47 | WIMPy | No, but you have a dynamic port which might make things more complicated. |
| 11:23.49 | hetii | good idea let me try |
| 11:28.56 | hetii | http://pastebin.com/xJR9XV3V |
| 11:30.16 | WIMPy | Looks like Asterisk actually replies, but probably to the wrong port. Why didn't we see that in the sip debug? |
| 11:32.24 | hetii | ohh it seams that the reason of it was that i use sip set debug peer 100600 instead sip set debug ip 127.0.0.1 |
| 11:33.20 | irroot | lol |
| 11:33.28 | irroot | been there done that |
| 11:36.08 | hetii | http://pastebin.com/pGMx1DpU |
| 11:38.16 | WIMPy | Ok, so it sends to the right port. Looks like t38modem is deaf. |
| 11:38.31 | hetii | deaf?:) |
| 11:38.59 | WIMPy | It doesn't hear the replies. |
| 11:39.27 | *** join/#asterisk irroot (~irroot@197.173.97.229) |
| 11:39.38 | *** join/#asterisk Tozz_ (~Tozz@hardwire.duocast.net) |
| 11:39.51 | hetii | irroot: http://pastebin.com/pGMx1DpU |
| 11:41.05 | irroot | ok looking better |
| 11:41.33 | irroot | is calling the nigie .... |
| 11:41.59 | irroot | only in south africa do we have a nigerian on speed dial ... even if not a nigerian |
| 11:42.52 | irroot | ok hetii the invite is coming in 100% right then asterisk responds with 401 [auth needed] |
| 11:43.13 | irroot | sends to port 44025 |
| 11:44.20 | irroot | check the nat / rport settings |
| 11:44.30 | irroot | in sip.conf |
| 11:44.53 | irroot | where it responds to port that it hears from |
| 11:45.36 | irroot | not the port configured |
| 11:45.49 | hetii | im not sure what you mean |
| 11:46.42 | WIMPy | But isn't that what should happen? |
| 11:46.43 | hetii | the nat=no |
| 11:46.48 | hetii | in general and extension |
| 11:47.38 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 11:48.21 | jsjc | hello I am getting an issue with this⦠Dial(DAHDI/7/${EXTEN},TK) is giving me error no timeout specified |
| 11:48.35 | jsjc | and if I put Dial(DAHDI/7/${EXTEN},,TK) it puts me the default timeout... |
| 11:48.46 | hetii | i found this: http://www.foriamroot.org/hylafax-6-0-debian-or-ubuntu-t38modem-1-0-asterisk-1-6/ |
| 11:48.50 | jsjc | is there anyway of telling to be forever with timeout? |
| 11:48.58 | hetii | there are some path maybe thats the reason ?:) |
| 11:49.38 | hetii | *patch |
| 11:51.20 | hetii | po co udar jak tam regipsy sa ?? |
| 11:51.35 | hetii | ups sorry :) wrong window:) |
| 11:52.26 | WIMPy | jsjc: 0? |
| 11:54.33 | jsjc | that is unlimited :)? |
| 11:55.50 | *** join/#asterisk jc319 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
| 11:55.53 | jc319 | hello |
| 11:55.55 | WIMPy | The doc syas it defaults to 136 years, if not specified. |
| 11:56.54 | jsjc | ok perfect |
| 11:57.11 | jsjc | 136 think will be fine. if I need 137 will try something else. |
| 11:57.33 | jsjc | app_dial.c:1849 dial_exec_full: Invalid timeout specified: '0'. Setting to infinite. |
| 11:57.36 | jsjc | good enough. |
| 11:57.59 | jsjc | pbx.c:4416 __ast_pbx_run: Timeout, but no rule 't' in context ' |
| 11:58.07 | jsjc | not infinte⦠but enough. |
| 11:58.30 | hetii | irroot: what you think about this patch that i found can you confirm that this can be a reason ? |
| 11:58.59 | hetii | in * 1.8 its look different so hard to say me if it was added. |
| 11:59.44 | jc319 | p3nguin: hello, are you here? |
| 12:01.41 | irroot | looking now was distracted |
| 12:02.56 | jc319 | I finally set up my asterisk with p3nguin's help, unfortunately I can neither have inbound calls nor outbound. I think there is something wrong with the ITSP, I will call them on Monday. Until then, are there any sip providers that would give me a free account with a dollar or less on it? or a trial account perhaps? |
| 12:04.12 | irroot | let me see the invite again |
| 12:04.28 | hetii | fine :> |
| 12:05.11 | irroot | ok it does not look like it applies as in your earlier trace the invite has no T.38 |
| 12:05.52 | irroot | and asterisk is responding with auth request |
| 12:06.19 | irroot | this is before this code is checked AFAIK |
| 12:06.31 | irroot | you need to respond with auth |
| 12:08.05 | hetii | is there anything that i can do? |
| 12:08.48 | irroot | the reply is going nowhere that is why it retrys from t38modem side |
| 12:10.19 | hetii | you man that port in Contact: <sip:100600@127.0.0.1:44025> is wrong ? |
| 12:13.28 | irroot | try set the contact to 6060 |
| 12:13.51 | irroot | it may be needed in the registrar opt |
| 12:24.56 | hetii | irroot: http://pastebin.com/WrgeLrWq |
| 12:26.56 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:7c23:92b3:7e50:df74) |
| 12:26.57 | irroot | intresting |
| 12:27.01 | irroot | looks good |
| 12:27.08 | irroot | 2 way traffic |
| 12:27.45 | hetii | but still SIP/2.0 401 Unauthorized |
| 12:28.00 | irroot | now its now a case of getting auth going propper |
| 12:28.18 | irroot | proxy auth and user auth options ?? try with both |
| 12:28.40 | irroot | now that you see it you can mess with it till it sticks |
| 12:30.54 | hetii | to be onest i don`t know what else i can change |
| 12:31.18 | hetii | i try already with/withour --sip-proxy option |
| 12:31.43 | irroot | i dont know why the auth is not working right |
| 12:32.09 | irroot | it should reply with 401 and then the t38modem should send a new invite with username/pw |
| 12:32.23 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 12:32.54 | hetii | hmm |
| 12:35.22 | southskies | I am seeing sudden unidirectional audio for all IAX2 channels after a few hours of operation. Thoughts? |
| 12:36.10 | southskies | I'll give you trial account |
| 12:53.14 | WIMPy | southskies: Your firewall acting up? |
| 13:04.27 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:09.25 | irroot | anyone using linux keen to go to v3 ?? |
| 13:11.04 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 13:20.14 | *** join/#asterisk Corydon76-home (one@c-69-137-80-31.hsd1.tn.comcast.net) |
| 13:20.14 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:30.54 | *** join/#asterisk mtnbkr (~mtnbkr@75-150-91-17-NewEngland.hfc.comcastbusiness.net) |
| 13:32.50 | mtnbkr | good morning everyone... Back in December I installed "asteriskNOW v1.7.1 with FreePBX v2.8.0.4" this installed Asterisk and freepbx with Centos 5.5... running a yum update shows that yum wants to update freepbx-2.7.0-6_centos5 to freepbx-2.7.0.7_centos5 - my FreePBX web gui shows FreePBX v2.9.0.6 (udates to freePBX since Dec have been via web gui) How can I properly maintain the OS without breaking and accidentally do |
| 13:32.50 | mtnbkr | wngrading my VoIP system |
| 13:40.26 | *** join/#asterisk golikwid|mac (~chrislees@64.45.192.10) |
| 14:03.02 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:03.44 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 14:13.47 | *** join/#asterisk vfabi (~fabi@194.247.164.231) |
| 14:15.17 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc7-pool12-2-0-cust219.15-1.cable.virginmedia.com) |
| 14:24.47 | *** join/#asterisk lost_soul (shawn@cpe-67-249-130-106.twcny.res.rr.com) |
| 14:25.49 | jc319 | I wanted to buy a voip account for tests only (inbound/outbound calls to pstn) with a DiD. It should work with asterisk and should be able to sign up for one month only - any recommendations? |
| 14:25.55 | jc319 | want* |
| 14:26.11 | mtnbkr | Sipstation I think will allow that. |
| 14:26.57 | *** join/#asterisk maxagaz (c4dbe572@gateway/web/freenode/ip.196.219.229.114) |
| 14:28.36 | jc319 | mtnbkr: The reason I'll have to buy a new one is that my current ITSP seems to send calls to voicemail directly ignoring my asterisk. I just don't want to buy a third one, if possible, so hope you don't mind me asking, are you using Sipstation with Asterisk currently? |
| 14:28.41 | maxagaz | hi |
| 14:28.58 | mtnbkr | $24/mo $ $1/mo each DID |
| 14:29.46 | maxagaz | my pbx and my sip app are in different vlans that can only communicate through the router, how can I make them communicate together |
| 14:29.46 | mtnbkr | Yes, I am using sipstation with an Asterisk/FreePBX install... Currently we only use the SIP truck for outbound calls so we do not tie up our incoming POTS line |
| 14:30.04 | maxagaz | creating routes ? if yes, how to do it ? |
| 14:30.16 | mtnbkr | maxagaz: you need a Layer3 device that can route between VLANS |
| 14:31.04 | maxagaz | mtnbkr: I can't do it without a device ? |
| 14:31.42 | mtnbkr | maxagaz: correct. |
| 14:32.06 | mtnbkr | dos your switch support L3 or is it just an L2 device? |
| 14:32.20 | mtnbkr | s/dos/does/ |
| 14:32.59 | maxagaz | mtnbkr: i don't know, it's a cisco catalyst 2960 |
| 14:35.13 | maxagaz | mtnbkr: i seems to be layer 2 device |
| 14:35.28 | jc319 | 2960s are L2 |
| 14:35.38 | jc319 | 3560s are L3 |
| 14:37.02 | mtnbkr | also, for an inexpensve solution, you can always install m0n0wall (or pfSense) on an embedded device like an ALIX board |
| 14:37.53 | *** join/#asterisk viro (viro@cpe-71-72-174-83.woh.res.rr.com) |
| 14:38.00 | viro | hi |
| 14:38.02 | mtnbkr | jc319: Thanks... was just looking at the datasheets for the 2960... it says " l2 with intelligent L2 through L4 services..." whatver THAT means.. |
| 14:38.02 | *** join/#asterisk golikwid|mac_ (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 14:38.06 | irroot | why not just use the linux box for routing |
| 14:38.11 | maxagaz | why isn't it possible to do it with routes ? |
| 14:38.24 | mtnbkr | maxagaz: because you don't have A ROUTER ;) |
| 14:38.33 | irroot | have all networks default to the linux box |
| 14:38.36 | maxagaz | mtnbkr: I have a router |
| 14:38.42 | irroot | and enable forwarding |
| 14:38.53 | irroot | a router that is on all vlans ?? |
| 14:39.02 | irroot | and is the defaut gw for all |
| 14:39.09 | maxagaz | irroot: yes, a router that is on all vlans |
| 14:39.24 | irroot | use vconfig on linux to config vlans |
| 14:39.29 | maxagaz | irroot: yes, it's the default gateway for all |
| 14:39.41 | irroot | then it should work ... |
| 14:40.28 | mtnbkr | irroot: you are getting too far ahead of him... If he has a router on all VLANs, then the router should already be routing between VLANS (unless it was not configured that way, or an ACL is preventing access to the voip VLAN from the device VLAN) |
| 14:40.39 | jc319 | mtnbkr: Many thanks Sipstation looks very good, also free software friendly (which is a plus for me). I'll look a bit more for another instant activation, no long-term commitment sip-to-pstn & DID service provider if can't find I'll go with sipstation. |
| 14:41.20 | mtnbkr | jc319: PLUS, if you use FreePBX, they have an auto-config module that takes the pain out of trying to configure the trunk if you are new to SIP trunks. |
| 14:41.27 | jc319 | I'm also looking for one with UK DID #s if possible because even though this is intended to be for tests, only for up to 30 days, you never know, if my current ITSP cant/wont fix this issues, I'll have to switch. |
| 14:41.30 | irroot | mtnbkr true ... other mistake i have seen is when the ip ranges overlap |
| 14:42.26 | maxagaz | mtnbkr, irroot, vlans have been configured manually |
| 14:42.43 | mtnbkr | maxagaz: can you decribe your network a little more? What is your router, is the uplink port of the 2960 configured as a general or trunk port, are all your VLANS tagged on that port to the router? etc |
| 14:43.49 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 14:45.58 | maxagaz | mtnbkr: the router is a centos machine, the uplink of the switch is configured as a trunk port (means all vlan go through this same gigabit port but don't communicate with each other, right?), and I don't know what means "tagged on that port" |
| 14:46.56 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 14:47.43 | mtnbkr | maxagaz: so, on the CENTOS box, the 802.1q module should be loaded, and each VLAN should have an IP address, and then what irroot was beginning to say is the direction you need to be going... |
| 14:48.01 | mtnbkr | maxagaz: on the centos box what does ifconfig show? pastbin it for us |
| 14:49.05 | mtnbkr | maxagaz: also show lsmod output |
| 14:50.00 | maxagaz | mtnbkr: I have 8021q in lsmod |
| 14:50.23 | mtnbkr | maxagaz: OK, so far, so good... now show ifconfig |
| 14:52.18 | mtnbkr | so, back to MY original question in here: |
| 14:52.27 | maxagaz | mtnbkr: ifconfig shows that each vlan has its own ip on virtual interface |
| 14:52.36 | mtnbkr | (repost) good morning everyone... Back in December I installed "asteriskNOW v1.7.1 with FreePBX v2.8.0.4" this installed Asterisk and freepbx with Centos 5.5... running a yum update shows that yum wants to update freepbx-2.7.0-6_centos5 to freepbx-2.7.0.7_centos5 - my FreePBX web gui shows FreePBX v2.9.0.6 (udates to freePBX since Dec have been via web gui) How can I properly maintain the OS without breaking and accide |
| 14:52.37 | mtnbkr | ntally downgrading my VoIP system |
| 14:53.19 | mtnbkr | maxagaz: can you ping both of the IPs from a device on the workstation VLAN, can you ping the VOIP server from the workstation? |
| 14:54.21 | mtnbkr | maxagaz: you may just need to enable IPv4 forwarding... |
| 14:55.02 | mtnbkr | what does cat /proc/sys/net/ipv4/conf/all/forwarding show? |
| 14:55.29 | maxagaz | mtnbkr: from the router I can ping both the pbx and the sip client's workstation, but the pbx and workstation can't ping each other |
| 14:55.38 | mtnbkr | maxagaz: also, what does iptables -L -n show? (maybe routing is enabled but you have firewall rules blocking traffic...) |
| 14:55.50 | mtnbkr | maxagaz: what does cat /proc/sys/net/ipv4/conf/all/forwarding show? |
| 14:56.01 | irroot | echo 1 > /proc/sys/net/ipv4/ip_forward |
| 14:56.04 | mtnbkr | echo 1 > /proc/sys/net/ipv4/conf/all/forwarding |
| 14:56.10 | mtnbkr | haha |
| 14:56.12 | maxagaz | mtnbkr: I have a lot of iptables (...) |
| 14:56.33 | maxagaz | forwarding is already enabled |
| 14:56.36 | irroot | iptables needs to be looked at esp nat / forward |
| 14:56.36 | maxagaz | on the router |
| 14:56.36 | mtnbkr | pastebin the /proc...forwarding output, then paste the rules to pastebin |
| 14:56.47 | irroot | going AFK |
| 14:58.54 | maxagaz | mtnbkr: cat /proc/.../ip_forward show 1 |
| 14:59.00 | maxagaz | shows |
| 14:59.15 | mtnbkr | maxagaz: so far, so good. |
| 14:59.20 | mtnbkr | now iptables |
| 15:00.29 | maxagaz | mtnbkr: iptables are very long |
| 15:00.43 | maxagaz | mtnbkr: and quite complex |
| 15:01.19 | maxagaz | mtnbkr: but I can't check logs in case something is blocking if you can guide me |
| 15:01.51 | *** join/#asterisk golikwid|mac_ (~chrislees@64.45.192.10) |
| 15:02.26 | mtnbkr | maxagaz: iptables -L -n | wgetpaste will paste them to a pastebin server and show you the URL to paste here. :) |
| 15:03.41 | maxagaz | mtnbkr: it's a bit sensitive, if you don't mind, I prefer not to paste it |
| 15:05.32 | mtnbkr | maxagaz: oh right. Hadn't thought about that. |
| 15:06.48 | mtnbkr | Then really,you need to check firewall logs and figure out how to add in some allow rules. How what the iptabl;es ruelest built? "Firewall builder" is an awesome gui for simplifying building iptables, ipf, pf, cisco PIX and a bunch of others. |
| 15:07.17 | irroot | wireshark / pcap / tcpdump are friends |
| 15:08.54 | *** join/#asterisk zapotek6 (~zapotek6@host151-5-dynamic.15-87-r.retail.telecomitalia.it) |
| 15:12.13 | maxagaz | mtnbkr: okay, but the client is trying to connect straight to the pbx, so shall I reconfigure it as well ? |
| 15:12.37 | mtnbkr | maxagaz: please explain that better... I am not sure I understand |
| 15:13.34 | maxagaz | mtnbkr: if the sip client connect has the ip of the pbx, it won't try to connect to the pbx through the router, right? |
| 15:13.50 | maxagaz | s/connect// |
| 15:14.45 | mtnbkr | maxagaz: the default gateway on the client is where it will send any packets on networks that it does not live on. |
| 15:15.50 | mtnbkr | maxagaz: so, the SIP client SHOULD point to the IP (or FQDN) of the voip server... the TCP/IP stack will take care of the rest... assuiming routes -n shows the IP of the Centos' IP on the workstation's VLAN which is what it will show if that workstation has INternet access ;) |
| 15:18.07 | *** join/#asterisk cneb3000-laptop (~cneb30000@02da1dc1.bb.sky.com) |
| 15:18.22 | cneb3000-laptop | howdy do |
| 15:18.50 | maxagaz | mtnbkr: so, you mean that I only need to add an iptable of port forwarding on the router ? forwarding a bunch of ports between the pbx and the workstation ? |
| 15:18.53 | irroot | lo there |
| 15:20.33 | cneb3000-laptop | guys.. i'm doing some research on predictive diallers. anyone have some material I can read? I'm reading predictive dialing fundamentals by aleksander szlam. Have also got some tech details on 2 dialers already out then. |
| 15:20.45 | cneb3000-laptop | slightly un-related to asterisk I know.. :) |
| 15:21.19 | cneb3000-laptop | particularly stuff about the algorythms involved, the architecture of the software. that sort of thing. |
| 15:21.41 | cneb3000-laptop | algorithm* |
| 15:22.07 | irroot | nice all i know about em is people who think they need them waste time ... |
| 15:22.49 | mtnbkr | maxagaz: not "port forwarding rules" but "allow rules" that allow your workstation VLAN (subnet) to communication to/from your voip server on the required ports |
| 15:22.59 | cneb3000-laptop | irroot: would you mind expanding on that a little? |
| 15:23.02 | irroot | most of them need a auto dialer not a predictive dialer due to the nature of there bussiness |
| 15:23.08 | cneb3000-laptop | ^^^ ahh i see |
| 15:23.12 | cneb3000-laptop | i completely agree |
| 15:23.38 | irroot | they ad hoc small centres with no plan / script or knowledge of what they want |
| 15:24.23 | cneb3000-laptop | i've seen predictive dialers make things worse for smaller centres, for sure. |
| 15:25.21 | maxagaz | mtnbkr: can you give me an example of allow rule ? |
| 15:25.23 | cneb3000-laptop | really just researching how dialers work in general. both in predictive, progressive and preview modes.. |
| 15:25.23 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-231.ips.direcpath.com) |
| 15:26.11 | mtnbkr | maxagaz: http://wiki.centos.org/HowTos/Network/IPTables |
| 15:28.52 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 15:28.56 | *** join/#asterisk lost_soul (shawn@cpe-67-249-130-106.twcny.res.rr.com) |
| 15:32.37 | jc319 | when you buy voip DID numbers & advertise them everywhere (printed marketing material, business card etc.) what happens if the ITSP goes out of business? Is there any way to save your #s? |
| 15:33.09 | p3nguin | With any luck you'll get wind of it and port your numbers out before they close. |
| 15:33.10 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 15:35.50 | p3nguin | Or solve the problem before it's a problem, and use a reputable ITSP with a good history. |
| 15:36.47 | *** join/#asterisk lost_soul (shawn@cpe-67-249-130-106.twcny.res.rr.com) |
| 15:38.29 | jc319 | p3nguin: Hello thanks for all the help yesterday. I decided to buy another sip trunk account for test purposes and if my current ITSP don't sort out the issues I'll switch. I came to the conclusion that they do not route my inbound calls to asterisk they just send it to VM. They're being nasty because the SIP indicator light is on in the webpanel & address is listed as my DID#@my-staticIP:5060. |
| 15:38.30 | jc319 | They even tell me my asterisk version. Re #s, yes will do once I get the basics sorted. |
| 15:40.51 | maxagaz | mtnbkr: thanks for your help |
| 15:42.47 | jc319 | Anyone uses voipvoip.com? |
| 15:43.37 | mtnbkr | maxagaz: no problem... I highly recommend installing and testing Fireweall builder. If your iptables ruleset is large, then this will greatly simply configuration and changes. |
| 15:45.07 | maxagaz | mtnbkr: I'm using a homemaid script to simplify firewall, I'm my boss is too "free software" to buy it, but I'll tell him about it |
| 15:45.15 | maxagaz | mtnbkr: thanks for this suggestion |
| 15:46.20 | maxagaz | s/I'm/I'm afraid/ |
| 15:48.15 | jc319 | p3nguin: Are you still using voip.ms? (in/out/both?) |
| 15:48.17 | mtnbkr | maxagaz: Firewall builder is open-source beer AND speach |
| 15:48.26 | mtnbkr | s/speach/speech/ |
| 15:50.03 | mtnbkr | irroot: BTW, I 'fixed' my FreePBX/ASterisk/Centos update dillema... the fix was to just exclude freepbx* in yum.conf since I continually update it via the web GUI, and excluding it will make yum update just skip any freepbx RPMs and will NOT downgrade me. |
| 15:52.04 | irroot | mtnbkr good thing |
| 15:52.14 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 15:56.23 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 16:00.55 | cneb3000-laptop | hmmm lots of dialers out there not being marketed very well. |
| 16:09.38 | p3nguin | jc319: I use VoIP.ms, yes. |
| 16:12.01 | *** join/#asterisk lost_soul (shawn@cpe-67-249-130-106.twcny.res.rr.com) |
| 16:16.02 | *** part/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-231.ips.direcpath.com) |
| 16:25.32 | jc319 | p3nguin: I have created a voip.ms account, ordered a DID # (takes up to 1 business days). Assuming you have more than one DID with voip.ms, do you have several sub-accounts? Do I set up a single sip trunk between my asterisk and voipms or do I register each number seperately? |
| 16:26.49 | *** join/#asterisk irroot (~irroot@41.54.210.35) |
| 16:37.12 | p3nguin | jc319: First of all, there is no such thing as a sip trunk. |
| 16:37.27 | p3nguin | I have one account for myself. I have all my DIDs on the account. |
| 16:38.07 | p3nguin | I provided a WORKING example of a voipms sip peer configuration with a register statement. Use it. |
| 16:38.50 | jsjc | i want to group to sip peers to make outgoing calls just 1 per peer. I been googling for a bit but cannot find much info any hints to start testing? |
| 16:42.15 | p3nguin | Asterisk makes the outgoing calls, not your phones. |
| 16:42.45 | p3nguin | Create your account. Take the account number and password and put them in my working example in place of my account number and password. |
| 16:46.20 | jc319 | I am sure it will work that way, just thinking about the next step - so the question is can I use the sub-accounts to have ITSP side seperate sub-entities (such as personal line1, personal line2, home line)? at the end of the month to see which sub-account costed what. I will try and see if it works but just wanted to get your opinion, to save time if this is a wrong approach. |
| 16:46.43 | p3nguin | Unless you have some very good reason, and I doubt that you do, to have more than one account per Asterisk, use my example. |
| 16:47.16 | *** join/#asterisk irroot (~irroot@197.174.102.199) |
| 16:47.31 | p3nguin | At this point you can make outbound calls. Add all your phone numbers to the account. Start making calls inbound. |
| 16:48.05 | p3nguin | For every new Asterisk system you deploy, create a new sub-account. |
| 16:48.10 | jc319 | We both have small businesses, (limited company with end-year accounts) so it'd help if we could distinguish business calls for company1, company2 and personal calls (home line). |
| 16:48.40 | p3nguin | Use your multiple lines on the phones. |
| 16:49.26 | jc319 | Yes I will use lines on the phones, how can I see what phone used how many pre-paid credits? (If possible I would like to track this on the ITSP side) |
| 16:49.34 | jc319 | Do I not need sub accounts for this? |
| 16:49.46 | WIMPy | I think the qwuestion was aboiut getting your beill sepperated by your DIDs as you get with your landline. |
| 16:49.49 | p3nguin | Or if you have a single-line phone, you can alter the caller ID to display some distinguishing marking. |
| 16:49.58 | p3nguin | One Asterisk, one account. |
| 16:50.19 | p3nguin | The phones do not use the ITSP; Asterisk uses the ITSP. |
| 16:50.28 | p3nguin | Your accounting will be done on Asterisk. |
| 16:50.50 | p3nguin | Remember than Asterisk is a B2BUA and not a proxy. |
| 16:51.01 | WIMPy | But Asterisk won't know the costs. |
| 16:51.22 | jc319 | Yes but I am very new to Asterisk (and running on a thin client) I can't guarantee it's health. (I like playing and testing new things). If I could pass this tracking finances task to the ITSP, it would not matter what happens to Asterisk (software problem/reinstallation/hardware issues etc.) |
| 16:51.25 | p3nguin | They won't unless you employ some type of billing software, which is outside the scope of this channel. |
| 16:51.49 | p3nguin | s/They/Asterisk/ |
| 16:52.05 | jc319 | (Plus this compact flash memory card which I use for read-write operations could die any moment from what I heard) |
| 16:53.04 | p3nguin | The ITSP does CDR (call detail/data records) based on phone numbers. Stop trying to complicate the process. |
| 16:53.05 | WIMPy | If you're on flash storage, tmpfs is a must. Best to use it read-only. |
| 16:53.28 | irroot | you need a TMS system Telephone Management System ... |
| 16:54.13 | *** join/#asterisk silkcut (silkcut@dan75-6-82-239-15-163.fbx.proxad.net) |
| 16:57.05 | *** join/#asterisk golikwid|mac (~chrislees@64.45.192.10) |
| 16:59.15 | *** join/#asterisk Tcalp (~Tcalp@mail.dejatechnologies.com) |
| 16:59.20 | jc319 | p3nguin: Not trying to complicate unnecessarily but I have ordered only 1 DID from voip.ms so far (as it was a test system initiall). Also DIDs do not matter when calling out, does it? I have 0 DIDs ready now, due to 1 business day provisioning time. Now seeing the control panel, voipms looks very advanced (compared to my current ITSP). *after* checking total cost of ownership I might cancel |
| 16:59.21 | jc319 | other ITSP and use this one, then I will have 3DIDs and I might change things then. At the moment what I am trying to do is setting up cisco phones so that depending on which 'line' button is pressed, the cost of the call should be added to the correct sub-account on voip.ms site. Just trying to see if this is possible with zero DIDs, asterisk and 1 voip.ms account (and 3 sub accounts). |
| 16:59.31 | jc319 | initally* |
| 17:00.24 | jc319 | WIMPy: I'll look into tmpfs, it is in my TODO list, thanks |
| 17:00.26 | p3nguin | I'm going to stick by my recommendation of "one Asterisk, one account." |
| 17:00.47 | *** join/#asterisk admin0 (~admin0@222.165.48.16) |
| 17:00.56 | jc319 | I assume it is very diffucult to set this up the other way then... |
| 17:01.05 | admin0 | hi .. what is wrong with this contenxt ? exten => _X.,1,Dial(SIP/vpsOUT/${EXTEN}) |
| 17:01.10 | admin0 | calls do not go to my sip provider |
| 17:01.12 | p3nguin | If you INSIST on using a peer per person, dial out via sub-account per phone. |
| 17:01.16 | admin0 | says extension not found in context |
| 17:01.18 | p3nguin | It's not difficult at all. |
| 17:01.41 | p3nguin | Create a sub account per phone that you have. |
| 17:01.44 | admin0 | i want all calls to go out to the vpsOUT peer |
| 17:01.56 | jc319 | If you don't mind explaining - why do you stick to the previous recommendation than? just trying to understand.. |
| 17:01.58 | WIMPy | admin0: Wrong context? |
| 17:02.27 | p3nguin | Create a peer entry in sip.conf for every sub-account that you create on the user portal. |
| 17:02.39 | admin0 | that is the only context in my extensions.conf and both sip friend and peer have the contenxt=vps |
| 17:02.43 | admin0 | basically same |
| 17:03.05 | Tcalp | anyone here familiar with Trixbox ? |
| 17:03.10 | p3nguin | You should not need register statements because registering is for inbound calls. You'll be using the sub-accounts for outgoing calls. |
| 17:04.16 | p3nguin | admin0: You should have only one sip peer definition for that ITSP. |
| 17:04.23 | WIMPy | Tcalp: Less likely. Try #trixbox. |
| 17:04.42 | p3nguin | Certainly not one that's type friend and another that's type peer. One entry of type=peer is enough. |
| 17:04.43 | Tcalp | WIMPy , yeah it's pretty dead in there ;) |
| 17:04.52 | p3nguin | Just like Trixbox! |
| 17:05.12 | Tcalp | I just need to add a trunk / inbound route / etc .. |
| 17:05.14 | WIMPy | Tcalp: The GUIs aren't very popular in here. |
| 17:05.33 | Tcalp | the person who setup our box no longer works here. |
| 17:05.34 | p3nguin | tcalp: Those are not Asterisk concepts, so you're in the wrong place to ask for support. |
| 17:05.53 | Tcalp | well I fugire the actual config files are the same ... (more or less) |
| 17:06.02 | irroot | Tcalp not at all |
| 17:06.07 | Tcalp | less the gui |
| 17:06.08 | Tcalp | oh |
| 17:06.25 | p3nguin | Since those are not Asterisk concepts, how would they be the same? |
| 17:06.26 | irroot | asterisk ships some example config files |
| 17:06.41 | p3nguin | And they are only examples. |
| 17:06.45 | WIMPy | Well, they are in theory, but that's unlikely to be of any practical help. |
| 17:09.41 | p3nguin | jc319: After you create your sub-account per phone, you can use a new variable within each phone's peer definition and reference that variable in the outbound dial command. |
| 17:10.28 | admin0 | WIMPy: my sip.conf and extensions.conf => http://pastebin.com/htrzym0K |
| 17:10.42 | p3nguin | Something like "setvar=OutPeer=sub1" for the first sub-account would work. |
| 17:11.58 | p3nguin | The vpsOUT peer does not have any credentials. I assume it's an ITSP, and they're going to require account information to be able to make calls. |
| 17:12.07 | irroot | my extensions.conf is insane it caters for every option concieved so far mostly via odbc |
| 17:13.14 | WIMPy | admin0: Are you sure the authentication of your phone succeeds? Maybe it's coming in as guest? Do you have guests allowed? |
| 17:13.24 | admin0 | p3nguin: in that server, its authenticated via IP address .. there is a trace running and there is no packets |
| 17:13.34 | admin0 | the error is: [Jun 11 09:00:00] NOTICE[18952]: chan_sip.c:20281 handle_request_invite: Call from '12389809' to extension '13038004561' rejected because extension not found in context 'vps'. |
| 17:13.40 | *** join/#asterisk golikwid|mac (~chrislees@207.30.30.130) |
| 17:13.51 | admin0 | WIMPy: its xlite softphone and status is registered |
| 17:13.57 | admin0 | sip show peers shows it as registered |
| 17:14.06 | p3nguin | but _X. should match '13038004561' |
| 17:14.15 | p3nguin | So make sure you reloaded your dialplan. |
| 17:14.17 | WIMPy | Registration has nothing to do with outgoing calls. |
| 17:14.18 | p3nguin | dialplan reload |
| 17:14.34 | WIMPy | But the message says it's the correct context. |
| 17:14.55 | p3nguin | You can confirm the extension with "dialplan show 13038004561@vps". |
| 17:15.55 | admin0 | ast*CLI> dialplan show 12389809@vps |
| 17:15.55 | admin0 | [ Context 'vps' created by 'pbx_config' ] |
| 17:15.56 | admin0 | <PROTECTED> |
| 17:16.42 | admin0 | i rebooted the box and now its working |
| 17:16.45 | p3nguin | That NOTICE should go away if you try again, since you've confirmed the extension@context. |
| 17:16.52 | admin0 | its working now |
| 17:17.00 | admin0 | i did asterisk reload |
| 17:17.01 | p3nguin | You probably forgot to run dialplan reload before. |
| 17:17.09 | admin0 | does it reload dialplan also ? |
| 17:17.19 | p3nguin | dialplan reload will reload the dial plan. |
| 17:18.34 | admin0 | aah |
| 17:18.40 | admin0 | thanks guys |
| 17:18.43 | admin0 | its working now |
| 17:25.58 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 17:28.07 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc7-pool12-2-0-cust219.15-1.cable.virginmedia.com) |
| 17:31.09 | *** join/#asterisk cneb3000 (~Ben@87.127.15.145) |
| 17:38.22 | jsjc | is theere anyway to limit concurrent calls per SIP trunk? |
| 17:38.51 | p3nguin | The SIP trunk is a lie. |
| 17:38.57 | jsjc | well a sip peer |
| 17:40.10 | irroot | the recomended approach is to use channel groups |
| 17:40.20 | irroot | call-limit depricated |
| 17:41.13 | p3nguin | jc319: This is probably how I would dial out using sub-accounts: http://pastebin.com/VAdXXvYh |
| 17:41.43 | Katty | hi |
| 17:47.33 | carrar | KattyRooooooooo |
| 17:47.34 | p3nguin | hi |
| 17:47.57 | p3nguin | What's for lunch today? I'm starvin'! |
| 17:48.04 | carrar | BEEF |
| 17:48.10 | p3nguin | I like beef. |
| 17:48.28 | carrar | BBQ |
| 17:48.33 | carrar | burgers |
| 17:48.50 | carrar | 1 pattie BEEF wiht 1 veggie burger on top of that |
| 17:48.57 | carrar | great mix |
| 17:49.05 | jsjc | iroot I have been looking into channel groups but cannot find an example that will understand or any info on it. |
| 17:49.22 | p3nguin | I BBQed some burgers the other day, so that's probably accurate. |
| 17:49.24 | carrar | jsjc |
| 17:49.30 | carrar | use GROUPCOUNT |
| 17:49.43 | carrar | and GROUP |
| 17:50.01 | carrar | and keep track of your channels used and then test everytime to see if the limit is over whatever you want |
| 17:50.09 | carrar | CAKE |
| 17:50.13 | carrar | next psl |
| 17:50.17 | carrar | pls |
| 17:50.35 | jsjc | jaja ok will investigate again then. |
| 17:50.55 | irroot | exten => _X.,n,SET(GROUP(out)=......) |
| 17:51.19 | carrar | SET(GROUP(out)=${ACCOUNTCODE}) |
| 17:51.22 | carrar | works well |
| 17:51.34 | carrar | if you use acount codes |
| 17:52.12 | irroot | exten => s,n,GotoIf($[${GROUP_COUNT(parent@out)} >= ${VLIM}]?s-OVERLIMIT) |
| 17:53.12 | jc319 | What happens to "setvar=externalCID=<value>" if I have no DIDs? |
| 17:53.18 | irroot | if i used parent above |
| 17:53.22 | carrar | GotoIf($[${GROUP_COUNT(${ACCOUNTCODE})} > $ACCOUNTCODEMAX}]?maxchan) |
| 17:53.29 | carrar | or something like that |
| 17:53.37 | carrar | err missing a { |
| 17:54.11 | p3nguin | Hmm. I have just been informed that we ate up all the burgers. |
| 17:54.15 | carrar | change those obvously |
| 17:54.23 | p3nguin | But I still have BRATWURST! |
| 17:57.32 | p3nguin | And, yes, they were cooked over charcoal. |
| 17:58.17 | p3nguin | My convenience grill (propane) needs a new burner, so I'm forced to do things right for a while. |
| 17:58.48 | irroot | need to break out some coals for lunch tommorow 4 shure |
| 17:59.40 | jsjc | mhn nice seems you guys eating good stuff⦠Nothing like meat over charcoal :) |
| 18:00.50 | p3nguin | I like to use my gas grill for ease of prep, but charcoal makes the food taste so much better. |
| 18:02.49 | p3nguin | With propane, I just have to turn it on, wait a few minutes, cook, shut it off. With charcoad, I have to dump the ash pan from the last cooking, pile up the charcoal and light it, wait 15 minutes, spread the charcoal around, cook, then wait for it to cool off to put it away. |
| 18:06.34 | irroot | maybe a lamb roast on the coals |
| 18:07.17 | p3nguin | When I cook on charcoal, I also get to cook corn on the cob. |
| 18:07.34 | irroot | p3nguin try a beer bread sometime |
| 18:07.52 | p3nguin | I've never baked in the BBQ before. |
| 18:07.54 | irroot | bottle beer some flour with herbs and spices |
| 18:08.18 | p3nguin | I've made beer bread in the oven, though. It didn't turn out very good. |
| 18:08.32 | irroot | been known to add some green herb containing THC |
| 18:08.45 | p3nguin | It was very heavy/dense. It had okay flavor, but that was about it. |
| 18:09.22 | irroot | yeah but good and easy when in the bush |
| 18:09.52 | p3nguin | I haven't been able to go camping for a long time. I need to do that soon. |
| 18:09.55 | *** join/#asterisk silkcut (silkcut@dan75-6-82-239-15-163.fbx.proxad.net) |
| 18:10.13 | irroot | join us at oppi-koppi |
| 18:11.01 | irroot | http://www.oppikoppi.co.za/ <- music festival on a "koppi" [hill] |
| 18:12.04 | irroot | i should apply for a liver transplant in advance :P |
| 18:12.28 | jc319 | Finally have the outbound calls working, thanks for all the help especilly to you p3nguin. |
| 18:12.59 | jc319 | Now there's a call in progress, if I reload sip & extensions would the ongoing call be dropped? |
| 18:13.01 | p3nguin | Are you using the sub-accounts and my suggested method to pick a peer for dialing? |
| 18:13.12 | p3nguin | No, it will not drop. |
| 18:13.48 | *** join/#asterisk golikwid|mac (~chrislees@207.30.30.130) |
| 18:13.55 | jc319 | No I just wanted to have a single working call before I do any more conf changes. I even removed all my previous config and copied voip.ms wiki example (which seems outdated) but at least it worked. I will improve on this now. |
| 18:14.56 | sezuan | Hi. Is it possible to change the jabber status of the asterisk jabber connection with AGI/AMI function? |
| 18:15.14 | jc319 | p3nguin: do you use g729? |
| 18:15.25 | p3nguin | Sometimes. |
| 18:17.13 | jc319 | Speed seems to be OK so far, do I need g729? voip.ms list only ulaw & g729 so if I only define ulaw, would it work all the time or depending on remote party's capability will I have connection problems? |
| 18:17.52 | p3nguin | The codec you select will affect nothing but the call between your Asterisk and VoIP.ms. |
| 18:18.28 | *** join/#asterisk rsp2k (18775bc7@gateway/web/freenode/ip.24.119.91.199) |
| 18:19.00 | p3nguin | But that call does weigh on a few things: bandwidth used per call, the call's quality, possible load on Asterisk's computer for transcoding... |
| 18:19.14 | p3nguin | s/call/codec/ |
| 18:20.15 | jc319 | Ok I'll leave only ulaw then, both in voipms definition and in cisco phone definitions... |
| 18:20.17 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 18:20.38 | p3nguin | For maximum bandwidth savings on a fair amount of calling, using g729 and an IAX2 trunk would be a good combination. |
| 18:21.05 | p3nguin | ulaw/alaw will give you the best sound quality on your Cisco phones. |
| 18:21.13 | irroot | of course if the phone is g729 then no transcoding cost |
| 18:21.26 | irroot | to the ITSP |
| 18:22.34 | p3nguin | I don't know how, but that just reminded me that I need to get fax working now that I have changed to a new system. |
| 18:23.32 | irroot | lol @ p3nguin i can do with out fax's for a while :P |
| 18:24.04 | p3nguin | I've been doing without faxes for a while already. I switched to a new system months ago. |
| 18:24.22 | irroot | that said you encouraged to test the t38 gw code :P |
| 18:24.41 | p3nguin | Asterisk 1.4 doesn't do t.38 as far as I know. |
| 18:25.00 | p3nguin | I was using the free fax for asterisk module, before. |
| 18:25.04 | irroot | ah you on 1.4 there is a way of doing it |
| 18:25.27 | p3nguin | I tried hylafax and iaxmodem, but I had bad results. |
| 18:25.28 | irroot | but not pleasant i have 1.8 patch for it and will be in 1.10 |
| 18:25.57 | p3nguin | If I ever get a working SCCP channel driver for 1.8, I'll use 1.8. |
| 18:26.33 | irroot | sorry bout that |
| 18:26.35 | p3nguin | I deployed 1.8 on the new system and had good luck with everything but SCCP. That forced me back to 1.4 and chan_sccp-b. |
| 18:27.23 | p3nguin | I saw that the chan-sccp web site said they support 1.8.0, but when going to the sf project page, the downloads section does not reflect 1.8 at all. |
| 18:27.28 | p3nguin | And I wouldn't use 1.8.0 anyway. |
| 18:27.51 | irroot | yeah its had some problems |
| 18:28.14 | p3nguin | It didn't even support the google voice stuff that I would want to use if I used 1.8. |
| 18:28.30 | p3nguin | That didn't come in until, what, like 1.8.1.1? |
| 18:29.56 | irroot | is outta here |
| 18:29.58 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 18:30.03 | irroot | cheers all |
| 18:33.23 | jc319 | p3nguin: should the ITSP definition be friend or peer? IIRC your example had peer - on their wiki ITSP definition has friend. Which one should I use |
| 18:33.33 | p3nguin | peer is suitable |
| 18:34.06 | p3nguin | It's pretty rare that any ITSP has _good_ samples. |
| 18:34.55 | jc319 | they do not have this bit>> dtmfmode=rfc2833 should I keep it |
| 18:35.04 | p3nguin | I've never met an ITSP that is good at configuring Asterisk for the end user. |
| 18:35.05 | jc319 | they also do not have any deny/permits |
| 18:35.48 | p3nguin | I gave you my WORKING example configuration for VoIP.ms. Do with it what you see fit. |
| 18:36.29 | p3nguin | If you need me to paste it again, let me know. |
| 18:39.58 | jc319 | I have it saved, thanks. |
| 18:47.07 | jc319 | for 7960s with multiple lines, do I need more than one 'device/peer definition'? You said something like MAC_a and MAC_b, I tried to use that method, but perhaps incorrectly did MAC_1 & MAC_2 definitions for a single hardphone. When line2 button is pressed it is complaining about it [check_auth: username mismatch, have <001A6CA3696C_1>, digest has <001A6CA3696C_2>] |
| 18:48.19 | p3nguin | Here's how I would configure sip.conf for using sub-accounts for outbound dialing per phone: http://pastebin.com/MLXLtb6Z |
| 18:49.25 | wdoekes2 | you shouln't need multiple accounts to use multiple lines on a single phone, jc319 |
| 18:49.27 | p3nguin | If you want to configure more than one line with different "settings," you'll want a sip peer for each line. I use MAC-a, MAC-b, etc. |
| 18:49.44 | p3nguin | If you want to do different things for each line key, you do. |
| 18:49.57 | wdoekes2 | does he? |
| 18:50.09 | p3nguin | I'm pretty sure he does. |
| 18:50.12 | wdoekes2 | if you get the error above, you probably need match_auth_username |
| 18:50.29 | jc319 | but Asterisk still 'sees' the default (1st line's) value (see the notice above) |
| 18:50.34 | p3nguin | It seems like a SIP<MAC>.cnf configuration error to me. |
| 18:51.33 | jc319 | match_auth_username=yes ---- if available, match user entry using the 'username' field from the authentication line ; instead of the From: field. |
| 18:51.41 | jc319 | OK will test this now, cheers |
| 18:51.45 | p3nguin | Create two peers in sip.conf for the two lines on the phone. [001A6CA3696C-a] and [001A6CA3696C-b]. |
| 18:52.50 | p3nguin | Then in SIP001A6CA3696C.cnf, configure auth for line one as 001A6CA3696C-a and auth for line two as 001A6CA3696C-b. |
| 18:52.54 | jc319 | OK I did that already (rather than -a and -b I used _1 _2 - corresponding to line buttons) |
| 18:53.21 | p3nguin | that'll be fine. |
| 18:53.30 | jc319 | That is done as well. Did not wark. I'm adding matchauthusr in general section of sip.conf now |
| 19:05.38 | *** join/#asterisk l2trace99 (~bender@rrcs-71-43-104-238.se.biz.rr.com) |
| 19:05.41 | jc319 | Sending to 10.9.8.71:5060 (no NAT) Using INVITE request as basis request - 000ab747-a418001b-5094507a-618ad969@10.9.8.71 |
| 19:05.46 | jc319 | Found peer '000AB747A418_1' for '000AB747A418_2' from 10.9.8.71:50460 |
| 19:06.20 | jc319 | it is correctly configured in IP<MAC>.cnf & in sip.conf. also in sip.conf "match_auth_username=yes" set in general section. |
| 19:06.22 | jc319 | Any ideas? |
| 19:07.06 | *** join/#asterisk maxagaz (c4dbe572@gateway/web/freenode/ip.196.219.229.114) |
| 19:07.09 | maxagaz | hio |
| 19:07.11 | maxagaz | hi |
| 19:08.39 | wdoekes2 | jc319: configure a different port for _2 ? |
| 19:10.06 | jc319 | wdoekes2: Port? |
| 19:16.02 | *** join/#asterisk cneb3000-laptop (~cneb30000@02da1dc1.bb.sky.com) |
| 19:16.33 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:18.48 | *** join/#asterisk dr0ck (~dr0ck@nat/digium/x-oiuqpjwjdzrsztnb) |
| 19:20.35 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 19:21.11 | p3nguin | Did you make sure you've used insecure=port? |
| 19:21.20 | p3nguin | There's no reason to configure a different port. |
| 19:23.03 | p3nguin | I use one port for all phones and lines on those phones. |
| 19:23.16 | p3nguin | That being the standard SIP port of 5060. |
| 19:25.07 | jc319 | In ITSP definition I have this "insecure=port,invite". In peer definition nothing about this. |
| 19:25.32 | jc319 | phone peer defs are host=dynamic, port=5101 one and 5102 the other line on the same phone |
| 19:25.36 | p3nguin | There's your mistake. Next time, use the information I provide. |
| 19:25.51 | p3nguin | And don't change port stuff. |
| 19:26.30 | p3nguin | http://pastebin.com/VAdXXvYh |
| 19:26.54 | p3nguin | Wait, that's the wrong link... |
| 19:27.14 | p3nguin | http://pastebin.com/MLXLtb6Z |
| 19:27.29 | p3nguin | That's the one with the sip sub-accounts. |
| 19:28.16 | jc319 | Yes I have that one, I was checking the caller ID first then I will work on sub-accounts. I have that paste before, saved this 2nd one too thanks |
| 19:29.20 | p3nguin | To make a second line, duplicate the peer for the first line, then append the _1 and _2 to the names. |
| 19:30.15 | jc319 | your code has 'defaultuser' asterisk 1.8 is complaining about it has been updated to defaultuser, should I not use defaultuser? |
| 19:30.22 | p3nguin | I use 1.4. |
| 19:30.27 | p3nguin | Adjust as needed. |
| 19:31.05 | p3nguin | But the phones don't use username or defaultuser, so that's moot. |
| 19:31.18 | jc319 | ok noted. Can I ask why do you not use 1.8? |
| 19:31.33 | p3nguin | There isn't a suitable SCCP channel driver available. |
| 19:32.06 | *** join/#asterisk adnc (~adnc@unaffiliated/adnc) |
| 19:32.16 | *** join/#asterisk golikwid|mac_ (~chrislees@207.30.30.130) |
| 19:33.43 | cneb3000-laptop | soon to delve into the world of IAX(2). wish me luck everyone :) |
| 19:34.09 | p3nguin | Even though it's not difficult... good luck. |
| 19:34.50 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 19:36.24 | cneb3000-laptop | comforting words p3nguin. |
| 19:38.16 | JerJer | sccp not friendly anymore ? |
| 19:39.14 | p3nguin | The last time I tested chan_sccp-b on 1.8.something, it did not work. The time I tried chan_skinny in the same version, it barely worked at all, and that isn't good enough. |
| 19:41.22 | jc319 | p3nguin: Last two pastes helps a lot. Along the way I lost extensions totally (have had only MAC addresses) it was one of the questions and that's sorted now too. One question with regards to sub-account divisioning though; why do we need to register with _90 when we have _91, _92 and _93 to call out? is _90 for DID only and 91/92/93 outbound only? |
| 19:42.07 | p3nguin | Registration is ONLY used for calling TO a device (or in this case Asterisk). |
| 19:42.47 | jc319 | With _90 what do we call TO? |
| 19:43.15 | p3nguin | In my configuration, as per your request, I am using a single sub-account for inbound calling and as a fallback for outbound calling. I'm going to refer to this account as the primary account. _90 is the primary account in my configuration. |
| 19:44.08 | p3nguin | You could use the main account as the primary if you want. I don't use my main account for anything, and I only use sub accounts for my various Asterisk systems. |
| 19:44.27 | p3nguin | I also only use one account per asterisk, though, which is not how you wanted to do it. |
| 19:45.09 | p3nguin | If you want, you could route incoming calls per sub-account. |
| 19:46.01 | p3nguin | The configuration I provided was a modification of how I would normally configure it, based on your request to send calls outbound on individual accounts. |
| 19:46.56 | p3nguin | To be quite honest, I don't know why we're still working on this. This is a 30 minute job that we're on day two of. |
| 19:47.39 | p3nguin | I do understand why you had to switch from that other ITSP. They are apparently a pile of crap. |
| 19:49.44 | jc319 | Well first day it took me some time to understand contexts, and config file structure - obviously no expert still, but at least I can modify and test things much faster now. 2nd half of 1st day passed with trying to fix/find a workaround to 1st ITSPs no-user-side-IP-PBX limitation. |
| 19:52.34 | jc319 | And about this design, yes to be able to track pre-paid credit usage on a sub-account basis I want to implement the new design you've provided. (because of several reasons I want to track usage on the ITSP side. as I said I do not want to keep Asterisk as the production system, I should be able to change/edit/reinstall without thinking about backups/exporting data (usage stats) etc. it may |
| 19:52.34 | jc319 | not make sense to you but for me it will be very convenient to have a (in terms of financial tracking) dumb IP-PBX). Using sub-accounts for outbound calls, my goals are, each 7960 (and softphones later on) should be able to call out using any of the available 'lines' (as it stands 1 business line and 1 home-shared line for each 'device'). |
| 19:53.36 | p3nguin | You plan to take out Asterisk? How's that going to work? Will phones communicate directly with the ITSP? |
| 19:54.17 | jc319 | You're right about the 1st ITSP by the way, they may not be crap, because some of their pages have slight references to 'running own ip-pbx on your site' so I think they support this but with manual configuration by their admins. At the moment there's no admins and there's no access to help so it won't work. Plus after seeing voip.ms control panel, I think even if they fix current issues, |
| 19:54.18 | jc319 | it'd be better to switch. |
| 19:57.37 | jc319 | As I said earlier running on a thin client & on a compact flash card without any tempfs optimizations it might die any day. Plus I want to finalize the design and have a read-only config backed up somewhere. And ideally have an identical thin client as cold-spare. If current unit dies (with or without tempfs optimization, due to CF card or any other component) ideally I would be able to simply |
| 19:57.38 | jc319 | swap thin client 'server' units and keep going. (not for my home obviously but for any other future sites, I can do this even on the phone, ask someone to unplug all cables and plug in to the spare unit and press the power button. If all dynamic details (call log etc.) are recorded on ITSP side, I would lose only 5 minutes of down time, and any home/business running telephony on the system |
| 19:57.38 | jc319 | I will provide can afford that anyway. |
| 19:58.07 | carrar | what |
| 19:58.16 | carrar | Katty |
| 19:58.21 | Katty | hi |
| 19:58.23 | carrar | hi!! |
| 19:58.25 | Katty | i just woke up |
| 19:58.26 | carrar | Whats the word? |
| 19:58.28 | Katty | i was napping |
| 19:59.00 | carrar | Did you drool? |
| 19:59.08 | Katty | not that i am aware of |
| 19:59.10 | p3nguin | You know she did! |
| 19:59.33 | carrar | heh |
| 19:59.51 | carrar | IRCing on a Saturday is just wrong |
| 20:01.48 | p3nguin | Today is Saturday?! |
| 20:01.53 | Katty | why? iv'e nothing else to do |
| 20:02.05 | p3nguin | I thought it was Sunday. |
| 20:02.12 | p3nguin | This is wonderful news! |
| 20:02.13 | carrar | it is in Japan |
| 20:02.26 | Katty | nothing to do tmw either |
| 20:02.27 | p3nguin | I just got a three-day weekend. |
| 20:03.17 | p3nguin | Yesterday was Saturday, and now today is Saturday again. |
| 20:04.08 | Katty | and what are you going to do with yourself? |
| 20:04.39 | p3nguin | I have a funny feeling that I won't do anything useful with my extra Saturday. |
| 20:05.48 | Katty | useful or not...what are you going to do? |
| 20:05.54 | Katty | i've spent over half of my day in bed |
| 20:06.25 | carrar | and the other half? |
| 20:06.41 | Katty | probably take the dog for a walk at some point |
| 20:06.44 | Katty | then back to bed |
| 20:07.00 | Katty | just can't motivate myself to do anything, bit down. |
| 20:07.16 | carrar | Why |
| 20:07.19 | carrar | What is going on!! |
| 20:07.23 | Katty | carrar: /query -> |
| 20:08.58 | p3nguin | I slept in, didn't get up until almost 10:30. I've spent a few hours here while watching crappy shows on msnbc. I'm probably going to go to Walmart and get some of those new wonderful southwest chicken taquitos they have. I don't really have anything else that needs done that I know of. |
| 20:11.20 | *** join/#asterisk lost_soul (shawn@cpe-67-249-130-106.twcny.res.rr.com) |
| 20:32.35 | *** join/#asterisk magn3ts (~magn3ts@ip68-103-225-65.ks.ok.cox.net) |
| 20:32.49 | magn3ts | So if I wanted to do something like Google Voice, what all technology would I need in total>? |
| 20:33.06 | magn3ts | Like, how does one acquire a phone number and then how does one gain the ability to send and receive SMS through it. |
| 20:35.24 | p3nguin | Sign up for Google Voice. Choose a phone number. Use the web interface or some iPhone app to send/receive SMS. |
| 20:35.46 | magn3ts | But I don't want to use Google Voice. I want to do it independently. |
| 20:35.58 | p3nguin | (1532.49) <magn3ts> So if I wanted to do something like Google Voice, what all technology would I need in total>? |
| 20:36.05 | p3nguin | This statement threw me off. |
| 20:36.52 | carrar | You'd need everything they use |
| 20:36.57 | carrar | simple! |
| 20:37.32 | carrar | or something LIKE it |
| 20:37.47 | p3nguin | Apparently the key word there was LIKE. |
| 20:38.01 | p3nguin | I missed it. |
| 20:38.30 | p3nguin | Oh well. Not the first time, and certainly won't be the last. |
| 20:38.33 | magn3ts | Like there is Plivo, but to my understanding another component still has to provide the actual connectivity necessary to send the SMS, someone upstream and I don't know who that would be, or what their technology would be called to google for them. |
| 20:39.07 | p3nguin | How about finding an ILEC? |
| 20:39.24 | carrar | or becoming one |
| 20:39.25 | p3nguin | Maybe they can help you with the phone numbers and they might even know something about SMS. |
| 20:40.57 | p3nguin | One problem I almost always encounter with service providers, though, is that they have no clue about the services they provide. I honestly don't know how they can even provide the services. |
| 20:47.08 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
| 20:47.49 | *** join/#asterisk l2trace99 (~bender@rrcs-71-43-104-238.se.biz.rr.com) |
| 20:50.12 | carrar | You can use Syniverse to get SMS access |
| 20:55.09 | *** join/#asterisk lost_soul (shawn@cpe-67-249-130-106.twcny.res.rr.com) |
| 20:57.59 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 21:05.03 | *** join/#asterisk jhmichelle352 (~jhmichell@41.206.150.28) |
| 21:17.51 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 21:33.09 | *** join/#asterisk digiv (~mlhess@as2.si.umich.edu) |
| 21:35.16 | *** join/#asterisk mer_ge (~martin@chello062178156192.9.14.tuwien.teleweb.at) |
| 21:37.53 | jc319 | My patience run out, registered an Albany number (was on the top of the list). |
| 21:48.22 | *** join/#asterisk lost_soul (shawn@cpe-67-249-130-106.twcny.res.rr.com) |
| 21:51.04 | jc319 | p3nguin: one more question? |
| 21:56.44 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:01.29 | p3nguin | waiting |
| 22:02.29 | p3nguin | The numbers only cost 1 USD, so if you decide you don't like it, drop it and get another. |
| 22:02.47 | p3nguin | At least that's what they cost here. |
| 22:03.45 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 22:17.22 | p3nguin | You also do not need to purchase a phone number to be able to call outbound. |
| 22:33.23 | jc319 | Yes I noticed that but I want to test some incoming # handling tests. However I also noticed that if I don't have several DID numbers I can't effectively test what I have on mind. Anyway 1 test DID is better than none, I will test what I can. |
| 22:33.45 | jc319 | voip.ms gave me $4/pcm price, how do you get for $1? |
| 22:34.20 | jc319 | BTW question was about the extensions.conf but I got it now (I guess) about to finish my modifications and test. |
| 22:35.39 | jc319 | IIRC, yesterday you said using _. is not a good idea to handle incoming calls right? so how can I do something like this in a 'better' way? exten => _.,1,NoOp(shared phone number); exten => _.,n,Dial(SIP/201&SIP/202&SIP/203&SIP/211&SIP/212&SIP/213&SIP/221&SIP/222&SIP/223,30); (basic idea is all incoming calls ring all extensions) |
| 22:36.07 | WIMPy | _X. |
| 22:36.33 | WIMPy | Only everything that starts with a digit. |
| 22:37.10 | jc319 | Cool thanks WIMPy |
| 22:46.58 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 22:49.59 | p3nguin | Like I told you multiple times already: explicitly specify the phone numbers as the extensions. |
| 22:50.38 | p3nguin | "ring all extensions" <--- extensions don't ring, phones do. |
| 22:51.57 | p3nguin | Once you have more than one phone number coming inbound, you're going to have to do it anyway, so why not doing it correctly from the beginning? |
| 22:53.07 | p3nguin | You think you're cutting corners to "just get it going," but all you're doing it making more work for yourself by not doing it the way it really should be done. |
| 22:54.28 | *** part/#asterisk screenn (~screenn@178.151.86.196) |
| 22:54.28 | p3nguin | We'll put the handlebars on later... get the tires on this bicycle now so we can start pedaling. |
| 22:58.06 | jc319 | I manage getting confused about extension/internal#/device. I think I got it now, editing extensions before getting confused again. Thanks |
| 22:59.07 | p3nguin | You make the association between an extension and a device with the "exten =>" lines in extensions.conf. |
| 22:59.31 | p3nguin | If there is no Dial() command, then there's no relation between an extension and a device. |
| 23:00.35 | jc319 | On top of that cisco .cnf has several buttons with 'line' label in the config file, that has been another source of confusion for me.. |
| 23:01.57 | p3nguin | Because IP phone "lines" aren't connected to actual phone lines? |
| 23:02.35 | jc319 | exten => 5185335329,1,Dial(SIP/001A6CA3696C&SIP/212&SIP/213&SIP/000AB747A418&SIP/202&SIP/203,30);; Incoming call on Home Line rings these devices |
| 23:02.56 | jc319 | ok I will use this one, and it will ring all the SIP phones (soft/hard) I need I think |
| 23:03.09 | p3nguin | As long as those devices are configured in sip.conf, you should be good to go. |
| 23:04.08 | p3nguin | If you're using _1 or _2 on your MACs to specify line 1 and line 2, remember to add them in your Dial() command. |
| 23:04.25 | jc319 | the only problem is, with this config what do I put in .cnf so that cisco deskphone user can do an outbound call from line A or line B depending on button pressed. Now before we did this with MAC_a MAC_b method and several peer defitinions in sip.conf for each hardphone. however it did not seem to be working because it always goes to default one (1st one). |
| 23:04.37 | jc319 | Is this something you've used before and should work 100%? |
| 23:04.44 | p3nguin | Yes. |
| 23:04.52 | p3nguin | Let me get the cnf again. |
| 23:05.44 | jc319 | I wonder if it is because of my firmware? I have P0S3-8-12-00 |
| 23:06.20 | *** join/#asterisk golikwid|mac (~chrislees@207.30.30.130) |
| 23:06.22 | p3nguin | http://pastebin.com/Y2nSPFbT |
| 23:06.48 | p3nguin | 8.12 works... all except for the caller ID display bug, which you haven't encountered yet. |
| 23:07.07 | p3nguin | But I still prefer to use 8.11. |
| 23:07.28 | jc319 | Is there any other differences than caller ID display bug? |
| 23:08.11 | p3nguin | I'm sure there's something different. I can't imagine that Cisco took a working 8.11 and bugged it intentionally just to release it as 8.12. |
| 23:08.29 | p3nguin | They could have, I suppose. They often do some silly things. |
| 23:09.25 | p3nguin | Use my configs, where things should work, then worry about the firmware. |
| 23:18.44 | p3nguin | I'm going to run out and get a liter of vodka and some food. I'll be back to check on your progress in probably an hour. |
| 23:19.31 | WIMPy | Homework? |
| 23:30.36 | *** join/#asterisk andrei1089 (~andrei108@81.18.92.10) |
| 23:34.39 | andrei1089 | Hi, how can I see the ast_debug output from chan_mobile ? I have tried to run asterisk with "-cddddd", but the messages are not shown. thanks! |
| 23:40.00 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 23:56.48 | Katty | hi |