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| 00:49.38 | pabelanger | cVsup: Create a context for that port and either ignore the call, let it ring, or Answer then hangup, to clear the line |
| 00:49.50 | pabelanger | Otherwise, you'll have to talk with your telco and make in only outbound |
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| 01:18.40 | tyler_d | Can you turn off ACL for a specific user? |
| 01:19.12 | tyler_d | I am trying to configure a softphone from outside of our intranet and need to dissable ACL |
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| 02:14.06 | cVsup | somebody can help with dahdi and callerid? |
| 02:14.12 | cVsup | in incoming routes? |
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| 02:21.00 | kaldemar | tyler_d: yes. just put permit=0.0.0.0 under the user definition in sip.conf. |
| 02:23.17 | kaldemar | cVsup: callerid=asreceived in chan_dahdi.conf for the channel. incoming routes? are you using a GUI? |
| 02:26.25 | cVsup | kaldemar yes |
| 02:26.31 | cVsup | i live in brazil and not work |
| 02:27.19 | cVsup | the cidsignalling is dtmf |
| 02:27.53 | cVsup | i have conversor from dtmf to fsk |
| 02:27.59 | cVsup | but not work |
| 02:28.39 | kaldemar | how does is not work? |
| 02:30.28 | cVsup | http://pastebin.com/jbdsHXUC |
| 02:32.00 | coppice | brazil has one of the more brain dead caller ID schemes. it can't be battery powered |
| 02:32.53 | cVsup | coppice yes |
| 02:33.04 | cVsup | but exist some providers than not |
| 02:34.15 | cVsup | but what say if i use dtmf converter to fsk? |
| 02:36.34 | cVsup | somebody can help? |
| 02:38.14 | tyler_d | kaldemar: is there a way to allow more granular control; ie. allow only a single extention to no use ACL? |
| 02:41.34 | kaldemar | tyler_d: what do you mean by extention? extensions in extensions.conf or something else? |
| 02:42.00 | tyler_d | only a specific user can log in with ACL=yes |
| 02:42.02 | tyler_d | not everyone |
| 02:44.03 | kaldemar | ACL=yes? |
| 02:45.21 | kaldemar | you can define deny and permit for each user if you want to. |
| 02:46.16 | tyler_d | doesn't take when you place ACL=no in the users.conf |
| 02:47.28 | kaldemar | where did you come up with such a parameter? |
| 02:47.44 | kaldemar | ~users.conf |
| 02:47.44 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 02:48.03 | tyler_d | lmao |
| 02:48.15 | tyler_d | sip.conf |
| 02:48.19 | tyler_d | !sip.conf |
| 02:48.26 | tyler_d | ~sip.conf |
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| 03:09.10 | ChannelZ | grunts |
| 03:09.38 | cVsup | kaldemar i found one problem |
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| 03:12.09 | cVsup | check this paste http://pastebin.com/0Jg9diwD |
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| 03:24.17 | ruben23 | hi guys can i turnon ipatbles on an asterisk server any recommedndation firewaall policy or rules for asterisk security.. |
| 03:26.05 | pabelanger | ruben23: yes, I'd recommend any public facing Asterisk box have a firewall |
| 03:26.58 | ruben23 | pabelanger:do you have any rules.. in particular i can sued.. |
| 03:28.27 | pabelanger | ruben23: it complete depends on the services you provide |
| 03:28.43 | *** part/#asterisk tyler_d (~cthompson@142.68.10.166) |
| 03:29.33 | ruben23 | <PROTECTED> |
| 03:31.39 | pabelanger | firewalls concepts are independent of Asterisk. Limit your IP address an port access to people who only need to access the box. By default deny everything, and only open it to people / clients that need it |
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| 04:20.24 | ruben23 | guys any idea how to block failed registration on asterisk..? the ip being used to register.. |
| 04:21.32 | teathsch | ruben23: use iptables, they won't stop though, so hopefully your network can handle it |
| 04:21.46 | teathsch | abuse complaints rarely work, but you can try that |
| 04:22.33 | teathsch | if it's not overwhelming your network though, iptables will prevent it from screwing up the asterisk process |
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| 04:28.57 | ruben23 | <PROTECTED> |
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| 04:29.39 | Dovid | Fail2Ban is real good for blocking it |
| 04:30.17 | teathsch | if it is completely unbearable, there's a script out there that will crash their process.. it didn't work when i tried it, but it's worth a shot if you get desperate |
| 04:30.38 | teathsch | sipvicious i think it was called |
| 04:31.25 | teathsch | yeah.. Fail2Ban reads your logs and decides.. failed regs => malicious and then runs iptables to block them |
| 04:31.33 | ruben23 | <PROTECTED> |
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| 04:37.46 | atan | Does 1.8 support realtime SIp peers w/MWI+NAT? |
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| 05:23.52 | ruben23 | hi guys by default permit and deny rule on sip.conf are not set right..? i need to add it..? |
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| 05:45.29 | p3nguin | I'm trying to deploy Asterisk 1.8.3.2 and I've having some issues. I've got SIP working like it's supposed to work, but I need chan_skinny to work correctly. Well, it doesn't work correctly, and I can't find any documentation about it, and that includes wiki.asterisk.org. |
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| 05:58.59 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 06:05.58 | p3nguin | If I configure skinny.conf as described in the comments, things just don't work out. For example, the linelabels... those are all jacked up. If I set a speeddial with a hint, the phone never loads up and there's an error on the CLI. I really need chan_skinny to work since chan_sccp is half-assed and doesn't work on Asterisk 1.8. |
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| 07:07.45 | kleszcz | morning |
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| 07:09.27 | jacc0 | hi all |
| 07:09.36 | jacc0 | I got this information on a segfault: |
| 07:09.37 | jacc0 | http://pastebin.com/hHND55Tz |
| 07:09.52 | jacc0 | can anyone make anything of this? |
| 07:10.30 | jacc0 | looks like it has got something to do with sleepmode or something |
| 07:11.31 | jacc0 | anybody any clue? |
| 07:12.51 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:13.48 | Wiretap7 | jacc0, your disk isn't failing? |
| 07:14.14 | Wiretap7 | segfaults are usually a result of something being either compiled for the wrong architecture, installed on the wrong OS, or some kind of hardware failure |
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| 07:23.46 | *** join/#asterisk casix (~casix@73.135.17.95.dynamic.jazztel.es) |
| 07:23.53 | casix | hello |
| 07:33.28 | casix | anyone knows how can I match quotes " in a regexp (the : operator)? I'm using asterisk 1.4.26.2. Thanks |
| 07:35.41 | Wiretap7 | \" |
| 07:37.44 | *** join/#asterisk kwk (~kwk@carbon.gonicus.de) |
| 07:39.29 | casix | Wiretap7: no |
| 07:39.43 | casix | doesn't match... |
| 07:39.56 | casix | i have tried " \" \\" \\\" |
| 07:40.00 | casix | but nothig works |
| 07:43.22 | *** join/#asterisk teathsch (~chatzilla@108-73-146-32.lightspeed.irvnca.sbcglobal.net) |
| 07:43.45 | jacc0 | i see this strange error in message log : astobj2.c: bad magic number 0x0 for 0xc9e36b8 |
| 07:44.20 | jacc0 | I guees I should call david copperfield for magic numbers |
| 07:44.41 | jacc0 | anyone had this error before? |
| 07:50.44 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 08:04.18 | jkroon | hi all, the effect of iaxthreadcount and iaxmaxthreadcount? |
| 08:04.46 | jkroon | at what values should these be set to on a relatively busy system (around 25 to 30 concurrent IAX channels at any given point in time) ? |
| 08:05.39 | jkroon | I'm getting lots and lots of chan_iax2.c: Max retries exceeded to host 41.240.193.177 on IAX2/sfsmarketing-1772 (type = 6, subclass = 11, ts=449995, seqno=68) |
| 08:06.21 | jkroon | as soon as those messages start my IAX2 channels dies and I'm unable to make further use of iax until a full restart, also, the CLI stops responding so I can no longer attach to the CLI using asterisk -r |
| 08:06.57 | zkn | jkroon:i get those too and i have not yet figured out what can solve this, so that makes two of us looking for a solution to that |
| 08:07.06 | jkroon | zkn, shit. |
| 08:07.16 | jkroon | pardon the language. but that is not good news. |
| 08:08.39 | zkn | what i have noticed is that the server does not hangup the channel when this happens, the call will keep on running in the seftphone whereas the inbound call was actually hung up... |
| 08:09.00 | jkroon | indeed. |
| 08:09.27 | jkroon | seeing similar stuff. looks like it _might_ be related to maxfd limits, but that would imply that somewhere along the line fd's are being leaked. |
| 08:09.34 | jkroon | which doesn't explain why SIP keeps functioning. |
| 08:09.54 | jkroon | currently at 1024, takes about two weeks to pop. |
| 08:10.01 | jkroon | which is about infinitely too soon. |
| 08:10.27 | zkn | how did you come to those numbers? |
| 08:10.33 | jkroon | the 1024? |
| 08:10.36 | zkn | yea |
| 08:10.39 | jkroon | checked in /proc/$(pidof asterisk) |
| 08:10.45 | jkroon | the two weeks is from experience. |
| 08:11.08 | jkroon | didn't have this issue in earlier versions of asterisk, manifested itself only after my upgrade from 1.6.2.13 -> 1.6.2.16.2 |
| 08:11.54 | zkn | yes, i started experiencing this, too in later 1.6 series and it is also happening in 1.8 versions |
| 08:13.19 | zkn | when call count is low and calls are short then this doesnt' occur |
| 08:13.38 | wdoekes2 | casix: try matching against a variable that contains the " |
| 08:14.19 | jkroon | argh, i sit with guys that can sometimes sit on a single call for an hour+. |
| 08:14.39 | zkn | jkroon, same here..same here |
| 08:15.38 | jkroon | tested 1.6.17 yet? |
| 08:16.24 | jkroon | zkn, ? |
| 08:16.37 | casix | wdoekes2: is what I do |
| 08:16.43 | jkroon | zkn, the file you want in /proc/$(pidof asterisk) is called limits. |
| 08:16.46 | casix | but it doesn't match |
| 08:17.12 | jkroon | then do a ls on fd/ and use wc to count the number of files (or f=(fd/*); echo "${#f[@]}") |
| 08:17.27 | zkn | jkroon: i'm running Asterisk 1.6.2.17.2, and i have seen this issue there, still |
| 08:17.38 | jkroon | ok, so upgrading won't solve it. |
| 08:17.42 | jkroon | craaap |
| 08:17.57 | jkroon | less frequently or about the same? |
| 08:19.10 | zkn | summertime is approaching, so number of calls have dropped and also this problem has become less frequent, but it exists |
| 08:19.50 | jkroon | dynamic IPs in your mix? |
| 08:20.10 | casix | wdoekes2: http://pastebin.com/rpxCWsBy |
| 08:20.28 | zkn | jkroon, for IAX2 no dynamic IPs |
| 08:20.31 | jkroon | zkn, also, have you managed to locate anybody else with the same problem? |
| 08:20.41 | jkroon | cool, that eliminates the other concern I had regarding this. |
| 08:21.00 | jkroon | bug reports/any other references? |
| 08:21.19 | zkn | i have not been able to do extensive research on this yet |
| 08:21.33 | zkn | been asking the same thing here for a cpl of times, no response |
| 08:21.39 | jkroon | cool. i'm just starting on it now, but i need to get it fixed. |
| 08:21.40 | zkn | i wish it was configuration issue and someone could tell what needs to be changed to fix this |
| 08:22.15 | jkroon | would you mind sending me your iax.conf file? Obviously with all actual IPs and secrets hashified or obscured in some way? |
| 08:22.50 | jkroon | i can then check what we have in common and see if there is anything from that set that might trigger it, also, the differences would (potentially) eliminate certain options. |
| 08:22.55 | zkn | i suppose I could |
| 08:23.24 | jkroon | that would be highly appreciated thanks. i sent you my email addy in pvt. |
| 08:23.56 | zkn | thanks, will send my info over in a bit |
| 08:24.05 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 08:29.25 | *** join/#asterisk iprouteth0 (ccf60469@gateway/web/freenode/ip.204.246.4.105) |
| 08:30.13 | iprouteth0 | does asterisk and dahdi still have issues on kernel 2.6.37? |
| 08:30.25 | jacc0 | do they? |
| 08:34.05 | p3nguin | I don't remember if I used to use 2.6.37 or not. |
| 08:34.39 | p3nguin | Now I'm using 2.6.32, so *shrug*. |
| 08:35.07 | iprouteth0 | there are a bunch of bugs posted for it. many seem like they may have been resolved though. My distro just added 2.6.37 sources though now into the stable branch |
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| 08:40.12 | p3nguin | I've been making a lot of changes lately and one of them was using an LTS kernel. |
| 08:45.10 | *** join/#asterisk casix (~casix@98.165.219.87.dynamic.jazztel.es) |
| 08:45.11 | casix | hello |
| 08:50.55 | jkroon | iprouteth0, see #gentoo-voip |
| 08:51.25 | jkroon | jacc0, dahdi-2.4.0 won't compile against 2.6.37 onwards. |
| 08:56.38 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:58.00 | eject_ck | Hi all, what's best way to deal with queues and FFA ? I have 2 fax licenses and need to be sure that no one outgoing fax call in queue will be rejected because of limit of sim. fax calls. How can I do that? |
| 09:20.47 | *** join/#asterisk ChB82 (~chris@2001:4cd8:1:0:221:70ff:fecd:7df1) |
| 09:31.21 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:35.27 | ChB82 | hello, can you tell me if and how it's possible to use digits in a queue (like ivr) in asterisk 1.4 ? |
| 09:39.16 | mzb | ChB82, I strongly suggest you increase your verbosity if you want an answer before Christmas |
| 09:39.17 | kaldemar | ChB82: see option "context" in the sample queues.conf. |
| 09:39.21 | mzb | ;) |
| 09:39.56 | mzb | *sigh* ;))) |
| 09:40.19 | mzb | great timing? |
| 09:40.37 | mzb | (re: the value of an act, etc) |
| 09:40.42 | mzb | nm |
| 09:54.41 | ChB82 | thank you very much! i will try it... :-) |
| 09:57.57 | ChB82 | okay...it works! :-D |
| 10:22.43 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 10:23.00 | zkn | Hi, has anyone had to deal with SIP stacked cell phones becoming UNREACHABLE to the server? this is happening only with cell phones, but not with computers running softpones in the the same WLAN..what to look at and how to eradicate the issue? |
| 10:25.49 | zkn | I have noticed better results when TCP has been used, but it still happens every now and then |
| 10:26.39 | *** join/#asterisk schmidts (~schmidts@2001:858:4:0:66b9:e8ff:fec3:ae88) |
| 10:26.43 | schmidts | hi there |
| 10:31.07 | kaldemar | zkn: find out what happens when the phone becomes unreachable. do they disappear from the WLAN? you could try decreasing the qualifyfreq for the phones if they drop out of the network between checks. |
| 10:48.07 | *** join/#asterisk m_tadeu (~quassel@89.180.156.4) |
| 10:51.06 | m_tadeu | hi...I need to receive simultaneous call from the same DID. my provider says I need to configure asterisk to enable this feature, but I don't seem to find what to do |
| 10:55.31 | *** join/#asterisk JonnyD_work (~Jon@cpe-071-075-036-057.carolina.res.rr.com) |
| 10:57.26 | kaldemar | m_tadeu: start by showing what goes on when you get more than one call. |
| 10:58.44 | jacc0 | @m_tadeu: change call-limit=1 in sip.conf iax.conf or your database |
| 11:00.07 | m_tadeu | kaldemar: nothing happens when I try the second call... jacc0, I'll take a look |
| 11:00.47 | jacc0 | @m_tadeu: also look at limitonpeers : http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers |
| 11:02.03 | *** part/#asterisk ChB82 (~chris@2001:4cd8:1:0:221:70ff:fecd:7df1) |
| 11:02.26 | kaldemar | m_tadeu: surely something happens in the CLI, either with verbosity or/and sip debug. |
| 11:09.24 | m_tadeu | jacc0: my mysql table doesn't have call-limit neither limitonpeer fields |
| 11:09.34 | m_tadeu | kaldemar: let me try with sip debug |
| 11:09.57 | jacc0 | <PROTECTED> |
| 11:11.21 | m_tadeu | kaldemar: nothing shows up on the second call |
| 11:11.39 | m_tadeu | jacc0: 1.8.3 |
| 11:11.56 | kaldemar | m_tadeu: if you're not getting a second call in the sip debug, then your provider is not sending one. there is no on/off feature in asterisk to enable multiple incoming calls, only different ways to prevent calls. |
| 11:12.05 | kaldemar | m_tadeu: nothing at all? no INVITE message? |
| 11:12.32 | m_tadeu | kaldemar: nothig... |
| 11:13.10 | jacc0 | <PROTECTED> |
| 11:13.10 | jacc0 | ./contrib/realtime/mysql/iaxfriends.sql |
| 11:13.28 | jacc0 | to see what collumns are available |
| 11:13.32 | kaldemar | m_tadeu: m_tadeu then you're not getting a call. |
| 11:13.46 | *** join/#asterisk radic (~radic@dslb-178-002-215-216.pools.arcor-ip.net) |
| 11:14.41 | jacc0 | <PROTECTED> |
| 11:14.45 | m_tadeu | kaldemar: I was suspecting that....but they replyed me saying I have to configure my trunk to receive silmutaneous calls |
| 11:16.27 | kaldemar | m_tadeu: tell them that you're not getting anything via SIP. it's not up to you. |
| 11:16.28 | m_tadeu | jacc0: I'll have to download that, since I'm using packages |
| 11:16.49 | m_tadeu | kaldemar: thanks a lot |
| 11:23.49 | *** join/#asterisk jeremy_g (jeremy_g@c-3ba0e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 11:30.08 | *** join/#asterisk ]loy[ (~nobody@95.72.26.143) |
| 11:31.29 | *** join/#asterisk iulhk (~iulhk@119.152.46.213) |
| 11:31.35 | iulhk | hi all |
| 11:32.18 | jacc0 | hi |
| 11:34.29 | iulhk | can we call recording if channel has been answered ? |
| 11:35.16 | jacc0 | @iulhk:mixmonitor |
| 11:36.01 | iulhk | ok thanx |
| 11:36.28 | iulhk | so during call its possible to record conversation ? |
| 11:38.54 | m_tadeu | jacc0: do you know what the default call-limit is? btw, those fields are not defined on the scripts either |
| 11:42.26 | kaldemar | iulhk: yes. if you want to start the recording during a call by pressing a key combination, see features.conf and "One Touch Record". |
| 11:43.01 | iulhk | thanxk |
| 11:47.11 | jacc0 | @m_tadeu: not sure what the default is, but it is more then 1 |
| 11:47.13 | kaldemar | m_tadeu: it is set to INT_MAX by default unless configured by the user. the size depends on your system, but for example on 32-bit systems it is 2147483647. should be enough for everybody. |
| 11:53.08 | m_tadeu | kaldemar: more then enough for me, for sure :) thanx guys |
| 12:00.36 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 12:00.50 | as001 | hi I have problem trying to register iax2 trunk |
| 12:01.02 | as001 | I get CAUSE : Registration Refused |
| 12:01.16 | as001 | following by CAUSE CODE : 29 |
| 12:03.30 | kaldemar | do you have access to the other end? |
| 12:03.54 | as001 | I have not it is winet.ch voip provider |
| 12:04.32 | as001 | I have register => number:password@server in iax2.conf and I am100% sure number and password are correct |
| 12:04.52 | as001 | number : password |
| 12:04.53 | kaldemar | what version of asterisk are you using? |
| 12:05.14 | as001 | 1.6.2.11 |
| 12:06.38 | kaldemar | did it work before? |
| 12:06.42 | as001 | Is there anything more to set ? What am I missing ? On winet i can see number which is checked like it is active. |
| 12:06.48 | as001 | It is new iax account |
| 12:08.01 | kaldemar | what did you mean by "number : password"? are there spaces around :? |
| 12:08.35 | kaldemar | wait a minute. iax2.conf? the configuration file is iax.conf. |
| 12:09.02 | as001 | no there is no space before and after : i did it because this chat type :p as this face |
| 12:09.15 | as001 | yes of course it is iax.conf |
| 12:11.21 | *** join/#asterisk J4zen (~J4zen@f108187.upc-f.chello.nl) |
| 12:13.58 | as001 | you can see my iax.conf here http://pastebin.ca/2045434 I tried with host=dynamic and host=ip |
| 12:14.35 | as001 | do I miss something ? |
| 12:15.57 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 12:16.31 | kaldemar | you're looking at the wrong place. the register statement is what matters. |
| 12:16.48 | as001 | whats wrong with it ? |
| 12:18.04 | as001 | its register => 00414457restofnumber:mypassword@sip04.winet.ch |
| 12:20.56 | kaldemar | since the cause code is 29, which means AST_CAUSE_FACILITY_REJECTED, looks very much like authentication failed. i.e. wither the username or secret is wrong. |
| 12:21.15 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl) |
| 12:21.51 | as001 | that username and secret i can see on my winet account |
| 12:22.55 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 12:23.39 | kaldemar | ask them what the problem might be. |
| 12:24.26 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:24.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:26.53 | as001 | ok |
| 12:27.06 | as001 | maybe they use other auth type.. |
| 12:27.22 | as001 | or my account is not active |
| 12:27.23 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
| 12:27.26 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 12:30.03 | *** join/#asterisk mheadd (~Adium@c-69-141-4-36.hsd1.de.comcast.net) |
| 12:30.17 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 12:35.15 | asterisk-learner | hello, if i want to add sounds files to asterisk, isn't it enough to just add them in asterisk/sounds ? |
| 12:35.34 | asterisk-learner | and when installing on a new system using make samples, they will get copied to /var/lib/asterisk/sounds ? |
| 12:36.03 | leifmadsen | asterisk-learner: /var/lib/asterisk/sounds/<lang>/ |
| 12:36.03 | leifmadsen | (in 1.6.2 and greater |
| 12:36.03 | leifmadsen | ) |
| 12:36.18 | leifmadsen | make samples copies sample files from configs/*.sample > /etc/asterisk/ |
| 12:36.20 | leifmadsen | nothing to do with sounds |
| 12:36.40 | leifmadsen | I usually make a subdir in /var/lib/asterisk/sounds/en/custom/ and then put my files there |
| 12:37.07 | asterisk-learner | so I have to cpy them manually on a new system ? |
| 12:37.09 | asterisk-learner | copy* |
| 12:37.19 | leifmadsen | no, make install copies the files over |
| 12:37.27 | leifmadsen | what you download and install is configured in menuselect |
| 12:37.43 | leifmadsen | including sound files |
| 12:37.45 | asterisk-learner | or is their a way to modify a conf or make file to do it in the instalation ? |
| 12:37.53 | leifmadsen | huh? |
| 12:38.07 | leifmadsen | there is always a way -- it's just code :) |
| 12:38.09 | asterisk-learner | yes i want to prepare an asterisk "package" that i will need to install on many mahcines |
| 12:38.17 | leifmadsen | ok, then you can do that |
| 12:38.20 | leifmadsen | you just need to build the package |
| 12:38.23 | asterisk-learner | so i thought i put them in asterisk/sounds |
| 12:38.44 | asterisk-learner | mmm what file(s) should i modify then ? |
| 12:39.06 | leifmadsen | look into how the configure build system works (it won't be trivial) |
| 12:39.18 | leifmadsen | or learn how to build RPM or DEB packages |
| 12:39.47 | leifmadsen | or write your own installation script that installs asterisk normally, then just copies the changes over top |
| 12:40.00 | leifmadsen | (the latter is the simplest method) |
| 12:40.16 | asterisk-learner | mm i prefer to use asterisk built in mechanism |
| 12:40.29 | asterisk-learner | i mean installer (make menuselect / make install) |
| 12:40.30 | leifmadsen | then you will need to learn how the build system works |
| 12:40.50 | leifmadsen | at the very least the menuselect system |
| 12:41.47 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:42.00 | asterisk-learner | which one is easier ? :-$ |
| 12:46.26 | asterisk-learner | in asterisk/sounds there is a Makefile, that contains this line : cp *.wav /var/lib/asterisk/sounds/ under "install:" |
| 12:46.44 | asterisk-learner | isn't this supposed to copy all .wav files ? |
| 12:54.46 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:54.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:59.05 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:02.34 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
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| 13:16.48 | *** join/#asterisk alami (~isy@unaffiliated/alami) |
| 13:17.02 | alami | hello, is there a german channel of Asterisk? |
| 13:18.22 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 13:20.03 | *** join/#asterisk cyford (~cyford@adsl-074-188-021-226.sip.asm.bellsouth.net) |
| 13:20.13 | *** join/#asterisk mheadd (~Adium@c-69-141-4-36.hsd1.de.comcast.net) |
| 13:20.23 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 13:21.04 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
| 13:21.24 | leifmadsen | alami: I do not see #asterisk-de sorry (there is #asterisk-br) |
| 13:22.15 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 13:23.01 | alami | leifmadsen: no problem :) i only have a problem with trunk, and german trunk are not like american trunks |
| 13:23.29 | leifmadsen | there should be some europeans here regardless |
| 13:25.32 | iprouteth0 | E1 vs T1? |
| 13:25.59 | SiNGLer | I missed the question, can it be repeated? |
| 13:26.12 | SiNGLer | (I am European) |
| 13:26.16 | alami | leifmadsen: i have set up an Asterisk server, and i want to test outside calls with a free trunks, but i havn't get it |
| 13:26.25 | leifmadsen | SiNGLer: there hasn't been a question asked yet |
| 13:26.37 | alami | that's it |
| 13:26.39 | leifmadsen | define: "free trunks" |
| 13:26.49 | alami | sipgate free trunk |
| 13:27.11 | leifmadsen | I don't understand how german SIP is different from north american sip |
| 13:27.21 | iprouteth0 | I shouldnt imagine that it is |
| 13:27.24 | alami | i have search and search and test but i still can't call and recive outside calls |
| 13:27.38 | leifmadsen | you'll need to provide a SIP trace of what is going on |
| 13:27.50 | iprouteth0 | alami: have you checked cli output? |
| 13:27.58 | alami | yes i have do it |
| 13:28.17 | alami | is there any other free and good sip trunk? |
| 13:28.39 | iprouteth0 | I never liked sipgate myself |
| 13:28.49 | alami | me tooo |
| 13:28.53 | alami | is too bad |
| 13:28.55 | iprouteth0 | I've used google voice, SIPgate, myipcomms.net |
| 13:29.02 | iprouteth0 | www.ipkall.com |
| 13:29.02 | alami | but i want only to test first you know |
| 13:29.05 | iprouteth0 | but I am also in US |
| 13:29.13 | iprouteth0 | fonosip was another one I think |
| 13:29.21 | alami | lol i wonder if they will work here |
| 13:29.30 | iprouteth0 | don't see why not |
| 13:29.33 | alami | but i will try these :D |
| 13:29.46 | alami | thanks a lot |
| 13:29.52 | henk | alami: how did you configure it? |
| 13:29.53 | cyford | hi, is there anyone here that have example agi survey scripts they can share? |
| 13:29.54 | iprouteth0 | The problem isnt where the SIP provider is, the problem is where the VoIP gateway is(where SIP server connects to PSTN) |
| 13:30.15 | iprouteth0 | So I guess it depends on where you are wanting to make calls to |
| 13:30.32 | alami | henk: i have do every thing i read it on the net |
| 13:30.52 | alami | from sip.conf to freepbx and trixbox everything i have do it |
| 13:31.23 | alami | i'm on germany, and i want to call german numbers |
| 13:31.39 | henk | alami: ah ok, so layer 8 problem... good luck |
| 13:32.13 | alami | thanks a lot |
| 13:32.33 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 13:35.43 | alami | iprouteth0: doese myipcomms.net offer also trunking? |
| 13:36.47 | iprouteth0 | alami: yes, though again I believe they are US based |
| 13:37.09 | alami | ok thanks |
| 13:37.37 | iprouteth0 | I signed up for a free sip DID/Trunk with them. Loved it, but as I now integrate directly with google voice, I dont use it so much anymore |
| 13:38.26 | iprouteth0 | I love asterisk+gtalk. Best solution yet for myself |
| 13:39.02 | cyford | is there anyone that would like to build one for an agi survey for $$ |
| 13:39.30 | iprouteth0 | agi survey?? |
| 13:40.20 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:40.32 | cyford | yes |
| 13:41.19 | cyford | one that starts after disconect, captures voice or options, and ads it to database |
| 13:46.08 | iprouteth0 | Do you mean like an IVR that calls back after a contact center call is completed to gather customer's opinion of the call experience? |
| 13:46.36 | henk | alami: hm, just tested the sipgate trunk, works perfectly, at least receiving calls... |
| 13:46.40 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:46.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:48.02 | cyford | well thats an option, but was thinking of having an option for the caller to opt in before they get transfered, or if the agent hang up before them give them a chance to take the survey again |
| 13:48.28 | alami | henk: rearly? are you on europ? |
| 13:48.41 | henk | alami: yes |
| 13:48.44 | alami | what do you use with Asterisk? |
| 13:49.03 | henk | sorry? |
| 13:49.47 | henk | yes, i use asterisk. was that the question? |
| 13:50.07 | alami | henk: can you tell me wich config have you use now? |
| 13:50.34 | henk | alami: just a simple register => with the account data from the sipgate trunk web panel and basic peer definition |
| 13:51.14 | alami | hmm |
| 13:51.54 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:52.03 | alami | henk: it will be very nice from you if you tell me on detaill what have do you done, or paste me that, and i will try only to change them |
| 13:52.14 | henk | http://pastie.org/1786892 |
| 13:52.20 | henk | was just preparing that ;) |
| 13:52.37 | *** join/#asterisk ariel_ (3fd6eca9@gateway/web/freenode/ip.63.214.236.169) |
| 13:52.45 | *** join/#asterisk coppice (~chatzilla@9.160.232.220.dyn.pacific.net.hk) |
| 13:53.08 | alami | lol cool, you are the best person on the world :D |
| 13:54.03 | alami | henk: you have done just that on sip.conf???? |
| 13:55.24 | henk | alami: oh, you barely know me, don't judge so fast... yup, that's all i've done. |
| 13:56.20 | alami | lol it's rearly crazy, because i have done the some thing |
| 13:58.05 | kaldemar | alami: if you have freepbx, it will break you hand-edited sip.conf. |
| 13:58.12 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 13:58.47 | alami | no no if i use freepbx i don't touch sip.conf |
| 13:59.41 | *** join/#asterisk dr__ (~duckz@78.96.110.191) |
| 14:07.42 | kaldemar | alami: then there is #freepbx for advice too. |
| 14:08.21 | alami | yes i know but i don't like it |
| 14:08.27 | alami | do you like freepbx? |
| 14:09.37 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 14:11.33 | iprouteth0 | I use the file editor in Asterisk GUI and avoid breaking my conf files |
| 14:14.15 | kuku | For the first time in 6 years working with asterisk, a new client has it running on a VM machine. The Vmware tools are installed. I moved it to a host with lots of free ram and fast scsi disks. The problem is the voice is still choppy. The choppines is even there when calling from extension to extension ( same location/switch ) so its not the sip TRUNK ( cbeyond ). Any suggestions ? |
| 14:16.24 | kaldemar | alami: no, and i don't use it. |
| 14:16.27 | Gugge | dont use a vmware guest for asterisk |
| 14:17.09 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 14:17.11 | alami | me too :) |
| 14:19.11 | *** join/#asterisk bchia (~chatzilla@nat/digium/x-saqkuxbjqwvvseyb) |
| 14:22.40 | iprouteth0 | say no to VMs |
| 14:27.24 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:29.40 | RickB17 | what causes a "SIP/2.0 403 Forbidden" I'm attempting to place a call between two asterisk boxes via SIP? |
| 14:32.20 | RickB17 | CSeq: 103 INVITE |
| 14:32.59 | psilikon | RickB17, probably password |
| 14:33.16 | RickB17 | psilkon: thanks i'll triple check them |
| 14:33.39 | RickB17 | both boxes show up successfully registered in "sip show registry" |
| 14:33.49 | RickB17 | and they show up as unmonitored peers in "sip show peers" |
| 14:34.08 | kuku | I fixed somehow that the voice is not choppy - but voicemail and tones are choppy. Any ideas? |
| 14:34.25 | kuku | Gugge: How so ? |
| 14:35.08 | leifmadsen | sounds like other VMs on the machine may be using too many resources |
| 14:35.24 | leifmadsen | you've also got a timing module enabled and internal_timing=yes enabled in asterisk.conf? |
| 14:40.44 | RickB17 | psilikon: i checked the passwords and they match |
| 14:40.53 | RickB17 | psilikon: i followed these directions, http://books.google.com/books?id=vtQxJ3oSm64C&pg=PA103&lpg=PA103&dq=send+call+to+another+asterisk+box&source=bl&ots=LWV7I_Jh67&sig=hZVPyysc9D_Jwbh7HRE-3_nkRYg&hl=en&ei=iFSkTebeDM_SiAK5wJnGCA&sa=X&oi=book_result&ct=result&resnum=9&ved=0CFcQ6AEwCA#v=onepage&q=send%20call%20to%20another%20asterisk%20box&f=false |
| 14:40.58 | RickB17 | opps thats kinda long |
| 14:41.18 | RickB17 | http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html |
| 14:41.27 | jacc0 | @rickb17: I noticed in my configuration that if sip/200 from server A is calling to server B and server B also as a sip account 200; you will have to use the password configured for sip account 200 on server B |
| 14:42.00 | jacc0 | to be able to make a call |
| 14:42.46 | jacc0 | strange imho |
| 14:43.35 | RickB17 | in my current setup i have a phone ext 200 registered to voice2. I am trying to call voice1. voice1 and voice2 both have the same sip record for ext 200, but it's only registered to voice2. |
| 14:44.25 | RickB17 | is that the same case you are mentioning? |
| 14:47.58 | Gugge | kuku: because vmware guests give crappy audio, dont use them for asterisk :) |
| 14:48.56 | jacc0 | @RickB17: yes |
| 14:50.21 | RickB17 | @jacc8: then how could i setup a simple trunk? I thought thats what this was doing? |
| 14:50.34 | jacc0 | @RickB17: try removing the 200 account or setting an empty password on voice 1 |
| 14:51.32 | RickB17 | @jacc8: same results |
| 14:51.55 | RickB17 | the password was the same of both voice1 and voice2 for this peer to begin with |
| 14:51.56 | jacc0 | @ RickB17: did you do a : sip reload on voice1? |
| 14:51.59 | RickB17 | yes |
| 14:52.17 | RickB17 | shouldn't it be only the asterisk server that needs to auth? |
| 14:52.38 | jacc0 | calling from voice2 to voice1 will use the trunk password |
| 14:52.57 | jacc0 | if you configure the same password for sip account 200 on both servers; try that |
| 14:52.59 | RickB17 | exten => 1300,1,Dial(SIP/voice1/${EXTEN}) |
| 14:53.18 | RickB17 | the sip account on both servers was the same from the start |
| 14:53.26 | RickB17 | i wanted to phone to be able to register to either or |
| 14:53.42 | jacc0 | and the 200 account useses the same password as the trunk? |
| 14:53.47 | *** join/#asterisk ms110 (~pil@203.88.130.74) |
| 14:53.47 | RickB17 | no |
| 14:53.54 | jacc0 | there is the problem |
| 14:54.32 | RickB17 | but all my sip users have different passwords, how woudl that work? |
| 14:54.40 | jacc0 | if you use the same password for the trunk and both sip 200 account it works for me |
| 14:54.48 | jacc0 | I don't know |
| 14:55.04 | jacc0 | I guess it's a bug |
| 14:55.22 | RickB17 | hmmm....strange |
| 14:55.29 | RickB17 | not sure that solution will work for me. |
| 14:55.33 | jacc0 | that was what I was thinking |
| 14:56.01 | jacc0 | try making a buggreport on issues.asterisk.org |
| 14:56.34 | RickB17 | pilsikon: do you have any suggestions? |
| 14:57.49 | jacc0 | I guess you are using asterisk 1.8.3.2? |
| 14:58.12 | *** join/#asterisk jared544 (5df60a48@gateway/web/freenode/ip.93.246.10.72) |
| 14:58.27 | jacc0 | sorry, I have 2 go now |
| 14:58.41 | RickB17 | 1.8.4 |
| 14:59.32 | jacc0 | okay, then it's a bug in both versions |
| 14:59.39 | jacc0 | ;) good luck with that |
| 14:59.46 | jared544 | Hello, I have install Asterisk Now 1.7.1 with the web-gui, but where I can find the username and password to login? |
| 15:00.15 | jacc0 | @jared544: manager.conf maybe |
| 15:00.18 | jacc0 | bye all |
| 15:01.07 | jared544 | oh thanks :) |
| 15:02.03 | RickB17 | when you enable sip debugging, is there an easy way to make that goto a log file? |
| 15:04.19 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 15:06.07 | RickB17 | nvm found this: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg126704.html |
| 15:08.42 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:09.36 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:15.19 | RickB17 | jacc0: it's def a password issue as stated above. I remove the secret from the sip.conf and it works. I'll have to review the passwords again and retry this. |
| 15:16.03 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
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| 15:17.01 | *** mode/#asterisk [+o BMJ] by ChanServ |
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| 15:32.00 | RickB17 | is there a good way through a dialplan to say "if the phone is registered call it, if not send it to a secondary server?" |
| 15:43.02 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 15:55.20 | Dovid | RickB17: Can you explaing a bit more ? You can check using chanisavail and make a decision based on that |
| 16:03.06 | *** join/#asterisk zkn (~zkn@82.131.93.186.cable.starman.ee) |
| 16:06.10 | *** join/#asterisk rosco (~rosco@eslpc28.epfl.ch) |
| 16:09.22 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 16:09.31 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 16:12.10 | leifmadsen | Dovid: ChanIsAvail() is not really a recommended method. You should be able to find something of use in the CHANNEL() function though |
| 16:12.46 | russellb | or DEVICE_STATE() |
| 16:13.12 | leifmadsen | oh ya! I just used that the other day too :) |
| 16:29.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:32.19 | ttpears | Been chasing this around for over a week, any clues: chan_dahdi.c: Write returned -1 (Resource temporarily unavailable) on channel |
| 16:32.37 | ttpears | Ruled out bad (TE122P) card |
| 16:34.15 | ttpears | It's repeating all the time, but will occasionally spew out that message into the log millions (literally) of times per minute, causing asterisk to come to a grinding halt |
| 16:35.51 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 16:36.46 | ruben23 | hi guys any help form this my asterisk server CLI is floodied with this display-------> http://pastebin.com/ehVCB0Bp |
| 16:40.25 | *** join/#asterisk m4xx (4b909aa5@gateway/web/freenode/ip.75.144.154.165) |
| 16:42.38 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-loeqfopawvhufmvt) |
| 16:42.42 | JonathanRose | #asterisk-dev |
| 16:43.05 | psilikon | ruben23, you're getting attacked |
| 16:43.31 | psilikon | ruben23, time to employ something like fail2ban and possibly country block |
| 16:43.46 | m4xx | i'm trying to connect my smartphone to my asterisk box from outside firewall, ive got the following settings: http://paste2.org/p/1359493 |
| 16:44.01 | psilikon | ruben23, http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk worked great for me |
| 16:44.07 | m4xx | my phone's sip client keep timing out when trying to register |
| 16:44.38 | m4xx | i've got nat enabled and ports 5060,10000-10500 forwared to my asterisk box |
| 16:45.50 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 16:46.11 | m4xx | anyone see any obvious errors and care to point me in the right direction |
| 16:47.46 | *** join/#asterisk [netman] (~netman@20.Red-80-39-52.staticIP.rima-tde.net) |
| 16:48.32 | ruben23 | psilikon: im using ubuntu server |
| 16:49.27 | p3nguin | I'm trying to deploy Asterisk 1.8.3.2 and I'm having some issues. I've got SIP and IAX2 operating like they are supposed to, but I need chan_skinny to work correctly or I can't use 1.8. But it doesn't work correctly, and I can't find any documentation on it, and that includes finding nothing about it on wiki.asterisk.org. |
| 16:50.37 | p3nguin | If I configure skinny.conf as described in the comments, things just don't work out. For example, the linelabels are all jacked up. If I set a speeddial with a hint, the phone never loads up and there's an error on the CLI. I really need chan_skinny to work since chan_sccp is half-assed and doesn't work on Asterisk 1.8. |
| 16:53.01 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 16:55.03 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:55.37 | m4xx | locally everything works fine |
| 16:58.23 | *** join/#asterisk bmg505 (~leon@196-209-99-25.dynamic.isadsl.co.za) |
| 16:59.00 | leifmadsen | p3nguin: unfortunately chan_skinny receives very little support |
| 16:59.01 | ruyo | Is there a way to check if a phone number is valid, through a SIP trunk, without making it ring? |
| 16:59.02 | leifmadsen | no one maintains that code |
| 16:59.19 | leifmadsen | ruyo: only if you have an API you can ask |
| 17:00.27 | m4xx | leifmadsen: any chance of you answering my question also? |
| 17:00.30 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:00.57 | ruyo | leifmadsen, so it depends entirely on the trunk? |
| 17:02.30 | leifmadsen | ruyo: it depends entirely on who you're interfacing with and if they provide some method for you to perform that lookup programmatically. There is no method that I'm aware of to determine if something is valid without actually calling it. |
| 17:03.06 | leifmadsen | m4xx: I don't see anything obviously wrong although you've only provided fairly vague information. |
| 17:03.29 | p3nguin | leifmadsen: I thought what qwell was trying to convince me of was that he was maintaining it. Maybe I misunderstood his message. |
| 17:03.32 | ruyo | Ok, that was what I suspected. Thank you leifmadsen. :) |
| 17:03.39 | m4xx | what else might i provide? |
| 17:04.07 | leifmadsen | p3nguin: he may have done some maintenance a while ago, but I do not believe he has been actively maintaining it for some time. |
| 17:04.26 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 17:04.58 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 17:05.15 | *** join/#asterisk CellKill (~kvirc@2001:470:c469:beef:45fe:6e2e:73f4:d474) |
| 17:06.27 | p3nguin | I really need some help with it. I do know that he insisted it is working, and that no one use chan_sccp-b because chan_skinny works correctly. |
| 17:07.02 | *** join/#asterisk jkprg (~jarda@ip-62-245-93-150.net.upcbroadband.cz) |
| 17:07.23 | p3nguin | I don't know what I can do to help fix it, because I am not a programmer and I certainly wouldn't understand how chan_skinny works internally. |
| 17:11.33 | Kobaz | p3nguin: i found at least three crashes in chan_skinny.. what asterisk version do you run on? |
| 17:11.52 | Kobaz | i posted up one of them in mantis |
| 17:11.56 | Kobaz | i need to do the others |
| 17:19.20 | p3nguin | This is 1.8.3.2. |
| 17:20.54 | Kobaz | yeah me too |
| 17:21.03 | Kobaz | does it crash for you on doing attended transfer? |
| 17:22.11 | ttpears | Has anyone done a 1.6 -> 1.8 upgrade, if so, was it painful? |
| 17:22.36 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 17:22.36 | ks3 | ttpears, Yes, and no, though YMMV |
| 17:23.10 | p3nguin | Crash? No, it doesn't work. |
| 17:23.39 | ttpears | ks3: Yea, figure as much, just curious, we just went from 1.2 -> 1.6, took months to complete & test |
| 17:23.49 | Kobaz | ttpears: define painfil |
| 17:23.54 | Kobaz | *ful |
| 17:24.09 | ttpears | Kobaz: Dialplan have to be re-written? |
| 17:24.10 | p3nguin | Well, if I adjust the config slightly so that it will work, I don't know if it would crash or not, because I only made one test call and didn't test features. |
| 17:24.24 | Kobaz | ttpears: most config is about the same, read the release notes |
| 17:24.47 | Kobaz | ttpears: as usual, test, test, and test, before deploying into production |
| 17:24.59 | leifmadsen | ttpears: it makes a big difference if you're talking about 1.6.0 vs 1.6.1 vs 1.6.2 -> 1.8 |
| 17:25.16 | leifmadsen | because all of those are separate branches -- just saying 1.6 doesn't really mean anything |
| 17:25.29 | Kobaz | it depends what you're using too |
| 17:25.29 | p3nguin | I wouldn't think the dial plan would need to be changed unless you're using apps or something that 1.6.something.whatever had and 1.8 lost. |
| 17:25.48 | Kobaz | i routinely switch between 1.6.0, 1.6.1, 1.6.2, 1.8, trunk |
| 17:25.50 | ttpears | leifmadsen: Good news for me then, 1.6.2 :-) |
| 17:25.51 | Kobaz | i have to change very little |
| 17:25.52 | leifmadsen | p3nguin: which shouldn't really be the case because we don't remove things anymore, even if they are deprecated |
| 17:26.14 | leifmadsen | ttpears: dialplan wise it should be trivial unless you're using old syntax |
| 17:26.22 | Kobaz | my agi stuff checks the asterisk version and sends either | or , depending on the version |
| 17:26.35 | Kobaz | some little things here and there, but mostly all my stuff works on any 1.6 and any 1.8 |
| 17:27.16 | iprouteth0 | I like 1.8 alot so far |
| 17:27.29 | Kobaz | 1.8 is cool, minus the crashes :( |
| 17:27.39 | ttpears | Thanks for the comments, I went through the changelogs the other day, it seemed trivial, but I was curious how others felt |
| 17:27.42 | iprouteth0 | mine's generally stable |
| 17:27.50 | iprouteth0 | small usage however |
| 17:28.03 | leifmadsen | also stable here for me |
| 17:28.09 | *** join/#asterisk JustinCampbell (~JustinCam@74-94-59-227-Philadelphia.hfc.comcastbusiness.net) |
| 17:28.14 | iprouteth0 | gtalk integration was flawless |
| 17:28.18 | ttpears | leifmadsen: large scale? |
| 17:28.19 | iprouteth0 | so easy to configure |
| 17:28.37 | itsbroken | Hello, I'm running asterisk 1.4.40 on debian. I have asterisk installed and running as asterisk user |
| 17:28.41 | iprouteth0 | It makes the system transparent to the end user |
| 17:29.15 | m4xx | are 5060,10000-20000 the only default ports needed to connect a sip client to a server? |
| 17:29.16 | itsbroken | What changes do I need to make to the init scripts so that it will start correctly at boot |
| 17:29.19 | iprouteth0 | How far can asterisk scale? |
| 17:29.29 | sezuan | How can I change the From line of the asterisks initial invite? at the moment it's something like "asterisk" <asterisk@ip> |
| 17:29.47 | p3nguin | I've got a client using 1.8.something and it is very stable. They aren't using chan_skinny, though. |
| 17:30.51 | p3nguin | m4xx: UDP 5060 and whatever UDP range found in rtp.conf |
| 17:31.00 | iprouteth0 | It's not uncommon that I speak with technicians from UK on our softswitches that I ask. ours is a strange configuration to me |
| 17:32.02 | m4xx | what is "what ever" ? |
| 17:32.17 | p3nguin | m4xx: whichever |
| 17:32.20 | p3nguin | m4xx: the one |
| 17:32.31 | m4xx | don't mean to be a pain, but i don't even get authentication failed |
| 17:32.34 | m4xx | it just times out =[ |
| 17:32.47 | p3nguin | m4xx: rtp.conf defines the start and end port for rtp. That is the range you use. |
| 17:33.13 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 17:33.21 | p3nguin | m4xx: If it is 10000-20000 in rtp.conf, then you use UDP 5060 and UDP 10000-20000. |
| 17:33.35 | p3nguin | m4xx: If it is 10000-12000 in rtp.conf, then you use UDP 5060 and UDP 10000-12000. |
| 17:33.58 | m4xx | that's what i've done =[ |
| 17:39.24 | m4xx | odd, i've opened 5000 to 105000 and can finally connect |
| 17:39.36 | m4xx | *10500 |
| 17:40.00 | m4xx | yet 5060,10000-10500 (the ports i defined in sip.conf and rtp.conf) alone don't work |
| 17:43.43 | p3nguin | You can't open only 10500 and expect any positive results. |
| 17:44.18 | m4xx | i had 5060 and 10000 to 10500 |
| 17:44.25 | m4xx | that counts as an even number |
| 17:44.26 | m4xx | no? |
| 17:44.50 | p3nguin | I have no idea what you're trying to ask. |
| 17:45.01 | p3nguin | You have to use the entire range as specified in rtp.conf, or a wider/larger range which entirely encompasses the range in rtp.conf. |
| 17:45.07 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 17:45.42 | p3nguin | And then, you must correctly configure Asterisk. |
| 17:46.02 | p3nguin | Asterisk is behind a NAT and you are forwarding those ports, yes? |
| 17:46.15 | m4xx | i have the following defined in rtp: rtpstart=10000 rtpend=10500 |
| 17:46.19 | m4xx | yes |
| 17:46.32 | m4xx | i have 5060, and 10000-10500 forwarded in nat |
| 17:46.53 | p3nguin | All UDP? |
| 17:46.58 | m4xx | both tcp and udp |
| 17:47.09 | p3nguin | You don't need TCP. |
| 17:47.44 | p3nguin | I suppose it might not cause critical failure to erroneously add TCP as well, though. |
| 17:48.22 | p3nguin | What explicit settings have you configured in Asterisk to make it work behind NAT? |
| 17:50.14 | m4xx | in sip.conf under [general] i have localnet, externalip set, under my extension i have nat=yes |
| 17:50.38 | m4xx | i also have canreinvite=no under [general] |
| 17:51.23 | p3nguin | There are no extensions in sip.conf. |
| 17:52.21 | p3nguin | (extensions are found in extensions.conf, surprisingly) |
| 17:54.09 | p3nguin | For SIP and NAT, you need to properly set values for nat, canreinvite, externhost or externip, and localnet. That's only for Asterisk being behind NAT. If you have phones or other devices which are behind a different NAT, you also configure values for nat and canreinvite within the peer definition. |
| 17:54.12 | m4xx | sorry, i have my terminology wrong |
| 17:54.50 | JerJer | [peer block] |
| 17:55.14 | JerJer | type=peer |
| 18:02.09 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 18:02.54 | Kobaz | anyways |
| 18:03.00 | Kobaz | don't know if you got it before my server went down |
| 18:03.00 | Kobaz | byt |
| 18:03.02 | Kobaz | but |
| 18:03.11 | Kobaz | leifmadsen: if you're not crashing asterisk you're not trying hard enough :P |
| 18:03.48 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 18:04.51 | p3nguin | I guess the server went down because of the extraneous s on anyway. |
| 18:05.06 | p3nguin | Could be something else, but I just don't know. |
| 18:06.29 | *** join/#asterisk jong2 (~chatzilla@63.224.204.153) |
| 18:06.38 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 18:08.18 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v001-053.mobile.uci.edu) |
| 18:20.11 | _Corey_ | p3nguin: Were you using chan_skinny with 1.6 w/o problems? |
| 18:22.01 | p3nguin | _corey_: Nope. I've never used any version in any 1.6.branch. |
| 18:22.27 | p3nguin | I use chan_sccp-b in 1.4 because chan_skinny just doesn't work right. |
| 18:22.37 | _Corey_ | aha... |
| 18:23.04 | _Corey_ | I happen to have a guy messing with it today in our lab for another project |
| 18:26.17 | *** join/#asterisk Xtrl (XtrlRulz@adsl-3-64-211.mia.bellsouth.net) |
| 18:27.17 | cyford | can anyone give me a clue as to why my pbx isnt accepting input from this survey script? http://fonality.com/trixbox/node/23377 |
| 18:28.06 | cyford | no voice or digits |
| 18:28.21 | Qwell | Because it's trixbox? |
| 18:29.14 | cyford | thought it uses the same dialing commands |
| 18:29.33 | Qwell | It doesn't even use the same software. |
| 18:29.51 | Qwell | They use some unnamed fork of Asterisk, that isn't anywhere near up to date. |
| 18:29.58 | benngard | guess people here are "should we say not not friendly to trixbox" ;) |
| 18:30.24 | benngard | me neither so i stand in line |
| 18:30.24 | cyford | ohh, it has freepbx instructions >>>> In FreePBX create a CUSTOM DESTINATION and in the "Custom Destination" box put in custom-customer-survey,s,1. Save those changes |
| 18:30.32 | Qwell | cyford: It's not FreePBX. |
| 18:31.11 | *** part/#asterisk mheadd (~Adium@c-69-141-4-36.hsd1.de.comcast.net) |
| 18:33.37 | p3nguin | _corey_: Messing with which thing on what branch? |
| 18:34.12 | _Corey_ | p3nguin: The various skinny channels on 1.6/1.8 |
| 18:34.35 | _Corey_ | (I think their results match yours) |
| 18:35.16 | p3nguin | If chan_skinny would work right, I'm ready to change to 1.8. That's the only thing holding me back. |
| 18:35.38 | Qwell | That and the looming deadline of 1.4 becoming EOL. |
| 18:35.42 | Qwell | in...EIGHT DAYS. |
| 18:35.52 | Qwell | 9 |
| 18:36.01 | Qwell | (also 1.6.2) |
| 18:36.14 | p3nguin | 1.4 doesn't go EOL for another year. |
| 18:36.37 | Qwell | meh. EOL enough |
| 18:36.49 | Qwell | It'll be security maintenance only. |
| 18:37.39 | p3nguin | _corey_: chan_sccp-b isn't compatible with 1.8, so it's not an option. chan_skinny isn't working right in 1.8, so I can't use it. My only choice is to continue using 1.4 and chan_sccp-b. |
| 18:40.19 | p3nguin | Beside the sccp problem, I'm otherwise "1.8 ready." |
| 18:40.42 | _Corey_ | :) Unfortunately, sccp support is what it is |
| 18:41.01 | _Corey_ | I stopped selling Cisco phones in 2005 |
| 18:41.47 | p3nguin | That seems like a good way to lose profits. No sales, no profits. |
| 18:42.03 | leifmadsen | people still by SCCP phones?! |
| 18:42.15 | leifmadsen | I'd imagine most SCCP enabled phones are coming from another existing system |
| 18:42.20 | _Corey_ | Well, I have a customer who inherited hundreds of them |
| 18:42.21 | leifmadsen | thus there are no sales anyways |
| 18:42.31 | leifmadsen | inherited == no money for joo! |
| 18:42.32 | _Corey_ | closed a bunch of locations and voila |
| 18:43.11 | p3nguin | wonders |
| 18:44.29 | _Corey_ | eh, it could be worse |
| 18:44.41 | _Corey_ | we could be talking mgcp or h323 or something |
| 18:44.44 | _Corey_ | ;) |
| 18:50.23 | m_tadeu | in sip show peer xxx, what is the meaning of "call limit: 0"? |
| 18:50.39 | *** join/#asterisk teathsch (~chatzilla@207.7.97.18) |
| 18:59.11 | leifmadsen | probably unlimited |
| 18:59.11 | *** join/#asterisk boazb (~boaz@bzq-82-80-219-90.red.bezeqint.net) |
| 18:59.11 | leifmadsen | m_tadeu: ^^^ |
| 18:59.48 | m_tadeu | leifmadsen: I thought so...but I wanted to be sure...can't find it anywhere |
| 19:02.56 | teathsch | what will happen if i send binary data on an iax2 text frame? will it come out the other end okay? |
| 19:04.35 | *** join/#asterisk zkn (~zkn@82.131.93.186.cable.starman.ee) |
| 19:12.50 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 19:13.04 | Aut0ExeC | when's the new asterisk book coming out? |
| 19:14.35 | carrar | be nice to read it while traveling |
| 19:14.51 | ttpears | Any idea if it's possible to use internal_timing with TE122P (and consequentially DAHDI)? |
| 19:14.53 | mocker | leifmadsen: You use Kamailio or OpenSIP? |
| 19:14.55 | Aut0ExeC | yeah i now |
| 19:15.01 | leifmadsen | mocker: no |
| 19:15.05 | mocker | neither? |
| 19:15.09 | leifmadsen | correct |
| 19:15.14 | Aut0ExeC | what does opensip do? |
| 19:15.16 | mocker | nods. |
| 19:15.16 | ttpears | Aut0ExeC: I thought it came out Friday? |
| 19:15.20 | leifmadsen | Aut0ExeC: google |
| 19:15.25 | Aut0ExeC | leifmadsen: k |
| 19:15.30 | Aut0ExeC | ttpears: nah |
| 19:16.10 | Aut0ExeC | I think leifmadsen is one of the authors too |
| 19:16.10 | mocker | leifmadsen: Wasn't asking for help/support btw, just wondering if you preferred one branch to another. |
| 19:16.24 | mocker | Seems like OpenSIPS got more activity. |
| 19:16.27 | leifmadsen | mocker: I haven't deployed either, so I can't answer that question |
| 19:16.42 | mocker | leifmadsen: Coo'. |
| 19:17.08 | leifmadsen | ttpears: the cookbook (ebook) is available, but Asterisk; The Definitive Guide is not quite ready yet -- soon though! Ebook of it MIGHT be available though, not entirely sure |
| 19:18.02 | ks3 | prefers OpenSIPS, but I wasn't asked... |
| 19:18.03 | leifmadsen | nice! http://www.amazon.com/gp/bestsellers/books/377896011/ref=pd_zg_hrsr_b_1_4_last |
| 19:18.38 | tzanger | oh stop gloating |
| 19:18.52 | leifmadsen | whistles innocently |
| 19:18.57 | mocker | ks3: I'll take it. :) |
| 19:18.59 | Aut0ExeC | leifmadsen: not released yet... |
| 19:19.05 | Aut0ExeC | leifmadsen: your a awesome author tho |
| 19:20.05 | ks3 | leifmadsen, I don't recall seeing your name on the first or second edition I've got. New, or am I just really getting forgetful? |
| 19:21.11 | leifmadsen | ks3: look again :) last name of the 3 |
| 19:21.26 | leifmadsen | Aut0ExeC: http://oreilly.com/catalog/9780596517342/ <-- electronic edition may be available (print edition is still at the printers) |
| 19:21.37 | leifmadsen | at least from going through the checkout it seems to be available |
| 19:22.14 | ks3 | leifmadsen, I'm just forgetful then... Guess I'll have to get the third edition, too. Might as well have them all! |
| 19:22.20 | ttpears | leifmadsen: I hope you're getting some of those profits! :-) |
| 19:22.36 | leifmadsen | ttpears: nope, Russell gets my cut |
| 19:23.18 | Aut0ExeC | leifmadsen: u do pbxing as a career? |
| 19:23.38 | leifmadsen | Aut0ExeC: google leifmadsen :) |
| 19:23.43 | Aut0ExeC | leifmadsen: my goal is to read your book to get l33t then start a solid career |
| 19:24.01 | Aut0ExeC | leifmadsen: possibly spanning various countries |
| 19:24.10 | ttpears | leifmadsen: heh, well I know here in the office they'll be two purchasers so ... thanks for your input in the book :-) |
| 19:25.25 | Aut0ExeC | leifmadsen: wow nice portfolio man |
| 19:25.51 | Aut0ExeC | leifmadsen: i aspire to be as l33t as you one day then pursure a career of similiar interest |
| 19:26.25 | leifmadsen | Aut0ExeC: lol I'm not l33t :) |
| 19:26.31 | leifmadsen | just ask tzanger |
| 19:26.39 | Aut0ExeC | leifmadsen: yes u are man.. i checked ur site |
| 19:26.44 | Aut0ExeC | leifmadsen: and u wrote a book |
| 19:33.15 | jaytee | not just a book, he co-wrote the * bible |
| 19:34.18 | jaytee | and soon he'll kick Julia Child's ass with the Asterisk Cookbook :-) lots of great recipes that don't require a pound and a half of butter. |
| 19:35.04 | benngard | :) |
| 19:35.30 | Aut0ExeC | lmao @ jaytee |
| 19:35.41 | Aut0ExeC | yeah hes the man |
| 19:36.08 | benngard | btw, any know of any good ipv6 ready phones, that supports colp |
| 19:36.29 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
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| 19:52.58 | leifmadsen | anyone run into an issue where a Polycom 650 that has multiple line buttons enabled just starts sending back a 603 Decline even though only 1 line is in use? |
| 19:55.05 | leifmadsen | wow reboot of the phone made it start working |
| 19:55.10 | leifmadsen | must be a bug in the firmware |
| 19:55.21 | leifmadsen | or the receptionist hit something accidentally |
| 19:57.54 | *** join/#asterisk sahX (~sahX@4.53.128.211) |
| 19:59.27 | leifmadsen | anyone here running asterisk 1.8.3.2? |
| 19:59.42 | leifmadsen | actually nevermind |
| 20:00.59 | omes | :) |
| 20:03.17 | leifmadsen | was going to ask for confirmation about a bug someone reported, but a note they left made it more clear what the problem was |
| 20:05.12 | Kobaz | for reels |
| 20:07.11 | jaytee | haven't used a Polycom 650 but I had a recently purchased 331 suddenly put it's lines into DND all by itself. A reboot fixed the issue but I thought it was "odd". |
| 20:08.08 | *** join/#asterisk radic (~radic@dslb-178-002-215-216.pools.arcor-ip.net) |
| 20:08.08 | leifmadsen | jaytee: aye... must be a bug in the firmware |
| 20:23.04 | *** join/#asterisk joshaidan (~brianj@24.109.210.41) |
| 20:24.17 | *** join/#asterisk CGMChris (~chris@74.143.228.142) |
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| 20:24.26 | CGMChris | res_config_mysql.c: MySQL RealTime: Failed to connect database server on (err 2002) <-- any idea how to turn this off? |
| 20:25.58 | Kobaz | don't load res_config_mysql |
| 20:26.53 | CGMChris | Kobaz: sounds reasonable enough, and easy to do with noload, I just want to make sure its not required. |
| 20:27.08 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 20:27.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:27.52 | Kobaz | it's not required if you're not using it |
| 20:27.56 | CGMChris | Kobaz: Do you know where I can find more info on what its used for? |
| 20:29.37 | Kobaz | it's used for storing configs in mysql |
| 20:29.50 | Kobaz | you can check voipwiki |
| 20:30.08 | CGMChris | Kobaz: So, unless I already know what it is and went out of my way to use it, I'm not using it? I'll double check voipwiki, thanks for your help. |
| 20:30.25 | Kobaz | voip-info.org actually |
| 20:30.33 | Kobaz | i never remember the name right |
| 20:30.51 | Kobaz | you're not using it especially if it cant connect to your db server |
| 20:31.50 | *** join/#asterisk cVsup (~cVsup@189.83.184.198) |
| 20:32.06 | cVsup | somebody can help with callerid before ring? |
| 20:35.14 | *** part/#asterisk BMJ (~bjohns@c-24-126-158-110.hsd1.ga.comcast.net) |
| 20:38.01 | *** join/#asterisk Yon (Yon@c80-217-241-69.bredband.comhem.se) |
| 20:42.50 | CGMChris | Kobaz: Disabling the module seems to have worked and didn't break cdr_mysql. Thanks! |
| 20:42.55 | *** join/#asterisk Schreiber1337 (cee4b465@gateway/web/freenode/ip.206.228.180.101) |
| 20:45.37 | Kobaz | yeah res_config_* is just for configuration |
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| 20:54.28 | cVsup | Kobaz, dtmf mode send callerid before ring. How can wait call for this? |
| 20:54.43 | cVsup | i found this |
| 20:54.43 | cVsup | http://www.spinics.net/lists/asterisk/msg122233.html |
| 21:00.14 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:14.47 | Kobaz | cVsup: does CALLERID() not do what you need? |
| 21:14.52 | Kobaz | core show function CALLERID |
| 21:15.14 | Kobaz | Set(CALLERID(num)=1234) Set(CALLERID(name)=Bob) |
| 21:17.55 | cVsup | Kobaz you read the thread? |
| 21:19.40 | Schreiber1337 | SIP hangs on 1.8.3.2 - 2 times a day or so... what do I need to setup to do a dump after a crash to figure out what is going on? |
| 21:21.51 | Schreiber1337 | I have to recompile with some debug options or something? |
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| 22:12.55 | fuse09 | hi there |
| 22:13.05 | fuse09 | i have a weird problem can you help me? |
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| 23:13.25 | cVsup | somebody can say what this? If i use cidsignalling=dtmf in chan_dahdi.conf my callerid not work, but if comment the line cidsignalling=dtmf and restart the asterisk, first call work callerid |
| 23:14.46 | cVsup | somebody developer can say about these functions? |
| 23:16.26 | cVsup | when these line show in asterisk log the callerid not work |
| 23:16.32 | cVsup | [Apr 13 02:14:04] VERBOSE[11206] chan_dahdi.c: -- Starting simple switch on 'DAHDI/1-1' |
| 23:16.33 | cVsup | [Apr 13 02:14:08] NOTICE[11206] chan_dahdi.c: Got event 18 (Ring Begin)... |
| 23:16.33 | cVsup | [Apr 13 02:14:09] NOTICE[11206] chan_dahdi.c: Got event 2 (Ring/Answered)... |
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| 23:42.34 | magicblaze007 | I'm updating my centos box...which updates the kernel...will that have any effect on my freepbx/asterisk installation? |
| 23:44.36 | Freeaqingme | a mere kernel update shouldnt |
| 23:45.04 | Freeaqingme | assuming it's from 2.6 to 2.6 |
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