| 00:04.23 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 00:20.59 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 00:22.00 | file | STUN is stunning. |
| 00:32.44 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 00:47.30 | *** join/#asterisk philipp64|laptop (~chatzilla@66-232-81-197.ctcweb.net) |
| 00:50.18 | ChannelZ | ha-HAA! |
| 00:51.34 | *** join/#asterisk mjordan (~mjordan@user-24-236-95-234.knology.net) |
| 00:51.34 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:59.23 | *** join/#asterisk philipp64|laptop (~chatzilla@66-232-81-197.ctcweb.net) |
| 01:13.30 | *** join/#asterisk eccr-phoenix (~phoenix@203.87.171.3) |
| 01:13.37 | eccr-phoenix | hi. |
| 01:13.49 | eccr-phoenix | what hardware is needed to use the software? |
| 01:19.25 | ChannelZ | a computer |
| 01:19.38 | ChannelZ | the rest depends on what you want to do |
| 01:19.53 | dj_hamsta | a router |
| 01:20.27 | *** join/#asterisk philipp64|laptop (~chatzilla@pbx2.redfish-solutions.com) |
| 01:26.15 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 01:30.14 | eccr-phoenix | and the phones...? RJ-45... |
| 01:36.30 | *** join/#asterisk philipp64|laptop (~chatzilla@66-232-81-197.ctcweb.net) |
| 01:41.30 | jpsharp | Either IP phones or analogue phones connected to analogue adapters. |
| 01:46.15 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 01:48.38 | eccr-phoenix | okay, thanks. |
| 01:54.43 | *** join/#asterisk greenwolf (4cb4d92f@gateway/web/freenode/ip.76.180.217.47) |
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| 02:27.04 | ChrisInSydney | hi, |
| 02:28.25 | ChrisInSydney | it looks like I may have to set up a system to accept unauthenticated IP calls that will need to route through a SIP provider. They will be comming from a limited range of IP addresses |
| 02:28.40 | ChrisInSydney | I feel really really uncomfortable about this |
| 02:29.48 | ChrisInSydney | I will still need to have authenticated calls from other IP addresses we have not blacklisted |
| 02:30.04 | ChrisInSydney | anyone have any suggestions ? |
| 02:30.43 | ChrisInSydney | maybe feed then through Kamailio ? |
| 02:33.24 | *** join/#asterisk mjordan (~mjordan@user-24-236-95-234.knology.net) |
| 02:33.24 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:39.06 | *** part/#asterisk mjordan (~mjordan@user-24-236-95-234.knology.net) |
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| 02:50.20 | carrar | ChrisInSydney, accept them via a VPN only |
| 02:50.49 | ChrisInSydney | cheers |
| 02:51.36 | ChrisInSydney | Then filter using SIPCHANINFO(peerip) etc ? |
| 02:52.26 | carrar | if it's unauthenticated, but it comes to you via a authenticated VPN why do you care what the IP is at that point? |
| 02:53.07 | ChrisInSydney | carrar: I still have authenticated end points from other public IP addresses |
| 02:53.14 | ChrisInSydney | so I cant block those |
| 02:53.32 | carrar | keep those on your public system or public interface |
| 02:54.08 | ChrisInSydney | ahh |
| 02:54.16 | ChrisInSydney | think ive got you |
| 02:54.26 | ChrisInSydney | set up static iP address peers |
| 02:54.42 | ChrisInSydney | make those on the VPN subnet |
| 02:54.45 | ChrisInSydney | ?? |
| 02:54.54 | carrar | yup |
| 02:55.00 | ChrisInSydney | K |
| 02:55.13 | ChrisInSydney | then I can contain them to their own context |
| 02:55.18 | ChrisInSydney | and subnet |
| 02:55.21 | ChrisInSydney | :-) |
| 02:55.26 | carrar | or dedicate a dynamic range of ip's that will be routed over the vpn |
| 02:55.47 | carrar | If nothing else keep the unauthenticated SIP traffic on the VPN |
| 02:55.48 | ChrisInSydney | never thought about it before as I have not needed IP only peers |
| 02:56.08 | ChrisInSydney | cool |
| 02:56.25 | ChrisInSydney | now to brush up on unauthenticated peers |
| 02:56.48 | ChrisInSydney | always done U/P plus IP erange masking |
| 02:56.53 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 02:57.02 | ChrisInSydney | cheers |
| 02:57.15 | ChrisInSydney | saves me filtering IP addresses Xp |
| 02:57.41 | carrar | and that doesn't solve security either |
| 02:57.46 | carrar | jsut filtering on ip |
| 02:57.55 | carrar | if you have unauthenticated public IP's |
| 02:58.12 | carrar | hence keeping that SIP session setup on the VPN |
| 02:58.14 | ChrisInSydney | I have a "broken" box that can not register properly behind a NAT box. I can load a VPN client on it and punch the calls that way |
| 02:58.50 | ChrisInSydney | a light OpenVPN coinfg should work |
| 02:58.52 | carrar | Or just fix your natting issues |
| 02:59.00 | *** join/#asterisk RubyRails (~justin@74.118.180.13) |
| 02:59.18 | ChrisInSydney | have done so with an older cut of Kamailio |
| 02:59.24 | ChrisInSydney | Milkfish on DD-WRT |
| 02:59.32 | ChrisInSydney | hackijng the .cfg as we type |
| 03:00.27 | ChrisInSydney | the issue is that the registration engin doesnt update the nonce values, therefore * treats it as a replay and ignores them as stale |
| 03:00.31 | ChrisInSydney | engine |
| 03:06.16 | ChrisInSydney | looks like its working :-) |
| 03:07.06 | ChrisInSydney | now to see if NAT works OK ? |
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| 05:38.54 | slicknick5181 | !dist |
| 05:40.10 | *** part/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
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| 06:17.42 | v0lZy | lo |
| 06:23.15 | ChannelZ | 127.0.0.1 |
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| 06:33.31 | ChrisInSydney | hey all |
| 06:34.39 | ChannelZ | ahoy |
| 06:35.57 | ChrisInSydney | got a curly one. More of a Kamailio / SIPs issue |
| 06:36.14 | ChrisInSydney | or anyone who is good with regex and sed |
| 06:36.36 | ChrisInSydney | I need to substitute the from address on a router thats doing a SIP proxy for me |
| 06:36.46 | *** join/#asterisk oyugik (~oyugik@41.84.143.52) |
| 06:37.33 | ChrisInSydney | Using a DD-WRT box with milkfish OpenSER running on it |
| 06:38.16 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 06:38.22 | ChrisInSydney | I thunk I found where the subst goes, but the regex is hurting my brain |
| 06:38.35 | ChrisInSydney | which, I may say is a little sensitive |
| 06:39.29 | ChrisInSydney | that aside, if I do substitute the from header so the call is accepted by the priprietory pbx I am hopting that the via: headers will let me get back to my box |
| 06:40.06 | ChrisInSydney | basically SIP service <-> Proxy <-> Proprietory IP PBX |
| 06:40.34 | ChrisInSydney | I can send calls out now, quite happily |
| 06:41.29 | ChrisInSydney | incoming I get a 503. which is" F&^* off you aint from aroun these ere parts ar ya" |
| 06:41.42 | ChrisInSydney | so I need the From to come from a LAN address |
| 06:41.52 | ChrisInSydney | usually they use Cisco CUBE |
| 06:43.15 | ChannelZ | What does the regex need to change, and change into? And what is actually doing it? |
| 06:44.59 | ChrisInSydney | I need it to change the from: <sip:number@domainname.org> to <sip:number@172.17.1.1> |
| 06:45.53 | ChrisInSydney | {SER|SIPS|Kamailio} has a subst which appears to work a bit like sed |
| 06:46.14 | bulkorok | <PROTECTED> |
| 06:46.59 | ChrisInSydney | subst('/^From:(.*)sip:(.+)@0*172.0*17.0*101[0-9.]*(.*)$/From:\1sip:\2@my.wan.ip.address\3/ig'); |
| 06:47.01 | ChrisInSydney | is whats there |
| 06:47.17 | ChrisInSydney | bulkrock: I have that. Aparantly I also need to swap the from |
| 06:47.25 | bulkorok | mmh |
| 06:47.35 | ChrisInSydney | thats for the to not from |
| 06:47.39 | ChrisInSydney | I think |
| 06:47.47 | bulkorok | could be |
| 06:48.03 | ChrisInSydney | the subst fixes inbound to outbound |
| 06:48.41 | ChrisInSydney | I cant quite work it out, so rather than cram it into my brain< i am simply typing on IRC and procrastinating, hoping for some inspiration |
| 06:48.50 | ChrisInSydney | or someone to do the hard stuff for me |
| 06:48.54 | ChrisInSydney | for free :-) |
| 06:49.10 | bulkorok | uac_replace_from() ?! |
| 06:49.28 | ChrisInSydney | cool |
| 06:49.35 | bulkorok | Replace in FROM header the display name and the URI part. |
| 06:49.52 | ChrisInSydney | need to replace the host |
| 06:49.57 | ChrisInSydney | the bit after the @ |
| 06:50.23 | bulkorok | http://www.kamailio.org/docs/modules/1.0.x/uac.html#AEN106 |
| 06:51.13 | ChrisInSydney | cheers |
| 06:51.17 | ChrisInSydney | I'll have a look |
| 06:51.29 | ChrisInSydney | no uac module on the router though |
| 06:51.41 | ChrisInSydney | no chance of integrating one either |
| 06:51.42 | bulkorok | :( |
| 06:51.48 | bulkorok | f** |
| 06:51.53 | ChrisInSydney | subst will have to do |
| 06:52.26 | ChrisInSydney | is that a regex ? |
| 06:52.31 | ChrisInSydney | :D |
| 06:52.35 | bulkorok | :) |
| 06:53.02 | bulkorok | http://www.amazon.com/Regular-Expression-Pocket-Reference-Expressions/dp/0596514271/ref=sr_1_fkmr1_1?ie=UTF8&qid=1339570340&sr=8-1-fkmr1&keywords=reilly+regular+ex ;-) |
| 06:53.13 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 06:53.28 | ChrisInSydney | http://www.regular-expressions.info + http://regexpal.com/ |
| 06:53.35 | ChrisInSydney | tutorial and tester |
| 06:53.45 | bulkorok | the electronic way :) |
| 06:54.23 | bulkorok | there's no way of loading tha uac module!? |
| 06:54.30 | ChrisInSydney | so I know what I am feeding before the @. I guess i can play with sed and see if my transformation works |
| 06:54.33 | ChrisInSydney | nup |
| 06:54.43 | ChrisInSydney | milkfish-dd on a remote router |
| 06:55.15 | ChrisInSydney | if this works, I am seriously going to look at porting a current version of either Kamailo or SIPS to broadcom chips |
| 06:55.30 | ChrisInSydney | there is some stuff in OpenWRT packages |
| 06:55.35 | ChrisInSydney | for their SDK |
| 06:55.38 | bulkorok | would choose Kamailio |
| 06:55.54 | ChrisInSydney | me too |
| 06:56.04 | ChrisInSydney | just trying to keep an open mind |
| 06:56.19 | ChrisInSydney | I cant seem to see any real difference |
| 06:56.41 | *** join/#asterisk greenwolf (~Penguin@cpe-76-180-217-47.buffalo.res.rr.com) |
| 06:56.56 | ChrisInSydney | SIPS was probably in front for a little while but since the SIP router project, I'm not so sure |
| 06:57.20 | ChrisInSydney | either way they are a little obscure. Not something your average punter is going to try to use |
| 06:57.46 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
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| 07:15.15 | wdoekes2 | ChrisInSydney: I ran a dual setup a while with both kamailio and opensips a while with the same (slightly m4'd) config.. but I had to give up kamailio when I needed dialoginfo_set |
| 07:15.57 | ChrisInSydney | ahh, and you wern't dedicated enough to attempt a port yourself ;-) |
| 07:15.58 | *** join/#asterisk itsurkg (~itsurkg@202.52.236.10) |
| 07:16.33 | wdoekes2 | indeed :) |
| 07:16.50 | ChrisInSydney | http://www.opensips.org/html/docs/modules/1.6.x/pua_dialoginfo.html ? |
| 07:17.07 | wdoekes2 | yes |
| 07:18.20 | ChrisInSydney | looks cool, more to procrastinate over :/ |
| 07:20.08 | itsurkg | hi all, is there any tool to monitor asterisk calls, duration via web? |
| 07:25.44 | *** join/#asterisk polysics (~polysics@ip-87-238-15-145.static.adsl.cheapnet.it) |
| 07:25.47 | polysics | hello! |
| 07:25.55 | polysics | where does one go buy international SIP numbers, please? |
| 07:26.06 | polysics | I need a number in Brazil, USA and China |
| 07:26.13 | polysics | all going to the same box |
| 07:26.48 | ChrisInSydney | DIDX I remember them from VUC http://vuc.me Can not comment on how good or bad they are |
| 07:28.40 | *** join/#asterisk beaver_rrr (beaver@crabhost.org) |
| 07:29.09 | *** join/#asterisk Diffen (~diffen@c-7476e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 07:33.20 | Diffen | Hello. I have a small problem here. Im trying to connect a FreePBX box to a Asterisk using a SIP-trunk. I have a register string that looks like this: register => login:password@asteriskip. That seems to be working. On my Asterisk i have setup a trunk with type=peer and the same login and password. I have set host=dnamic. When the Free PBX tries to registers my Asterisk box says no matching peer found. The login and p |
| 07:33.20 | Diffen | asswoard are correct (i have copied and pasted them) and the type is peer. What have i missed out? |
| 07:34.28 | kaldemar | Diffen: [login] |
| 07:34.45 | ChrisInSydney | Diffen: You are missing out on time with those important people who love you |
| 07:35.14 | ChrisInSydney | but you are here now, so dont give up, they'll be waiting for you tomorrow |
| 07:35.17 | ChrisInSydney | :D |
| 07:35.18 | Diffen | kaldemar: ? |
| 07:35.27 | Diffen | ChrisInSidney: true :D |
| 07:35.30 | beaver_rrr | how to redirect incoming call from number X according to status of last call to this number from my network? |
| 07:35.40 | ChrisInSydney | what SIP stuff are you seeing |
| 07:36.02 | beaver_rrr | im looking into AGI, but not sure if it is right thing to look into |
| 07:36.03 | ChrisInSydney | sip set debug IP <theIPaddress> |
| 07:36.17 | ChrisInSydney | have a look at what comes back |
| 07:36.27 | ChrisInSydney | have a look at both consoles |
| 07:36.49 | ChrisInSydney | see what you get when you do a sip reload on the registering box |
| 07:38.00 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 07:38.33 | Diffen | in asterisk: chan_sip.c:15933 handle_request_register: Registration from '<sip:username:password@ip-address>' failed for 'ip-address from freepbx' - No matching peer found. i dont have access to the console of the free pbx server |
| 07:38.37 | kaldemar | Diffen: the peer does not match because what you have inside square brackets in the peer definition differs from your "login". |
| 07:38.48 | Diffen | ok |
| 07:38.52 | Diffen | ill check |
| 07:39.07 | ChrisInSydney | that'll be it |
| 07:39.49 | raden | i just isntalled asterisk 10 |
| 07:39.56 | raden | on console I have no sip commands |
| 07:40.28 | kaldemar | raden: you don't have chan_sip loaded. |
| 07:40.39 | raden | wtf |
| 07:40.42 | kaldemar | raden: what does "module load chan_sip.so" say? |
| 07:41.04 | raden | starcom-debian-asterisk*CLI> module load chan_sip.so |
| 07:41.04 | raden | Unable to load module chan_sip.so |
| 07:41.04 | raden | Command 'module load chan_sip.so ' failed. |
| 07:41.04 | raden | SIP channel loading... |
| 07:41.09 | beaver_rrr | http://img.leprosorium.com/1484118 - how to achieve such redirect? agi script? |
| 07:41.34 | *** join/#asterisk Micc (~Mic@c-24-19-33-189.hsd1.wa.comcast.net) |
| 07:41.48 | Micc | what is the lib needed to have make menuconfig come up in color? |
| 07:42.26 | kaldemar | beaver_rrr: no agi needed, inspect the DIALSTATUS variable after app Dial and make your decisions based on the value. |
| 07:42.36 | beaver_rrr | thanks! |
| 07:42.43 | raden | WTF |
| 07:42.47 | raden | this sucks |
| 07:43.54 | kaldemar | raden: "core set verbose 10" and try again. maybe unload the module first. |
| 07:44.31 | raden | nope |
| 07:44.32 | raden | no go |
| 07:47.06 | raden | kaldemar, I'm not getting any specific error eiyther |
| 07:48.13 | raden | gah i should have used 1.8 |
| 07:49.34 | raden | Unable to load module chan_sip.so |
| 07:49.34 | raden | Command 'module load chan_sip.so' failed. |
| 07:49.34 | raden | SIP channel loading |
| 07:49.37 | raden | thats all i get |
| 07:54.11 | raden | kaldemar, is it not in make by default ? |
| 07:55.20 | raden | I love snap shots on vm's |
| 07:55.23 | raden | role it back LOL |
| 08:00.55 | kaldemar | raden: did you upgrade from an earlier version? |
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| 08:17.06 | raden | kaldemar, clean install |
| 08:17.18 | raden | kaldemar, trying a different asterisk version now |
| 08:17.24 | raden | gah id like to get some sleep someday |
| 08:21.46 | kaldemar | raden: do you have a sip.conf in your config dir? |
| 08:23.22 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
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| 08:41.35 | raden | kaldemar, ill have to look in a min installing 1.8 |
| 08:41.49 | raden | ill revert system to that install |
| 08:44.58 | raden | bla |
| 08:56.56 | *** join/#asterisk shadebob (~shadebob@41.142.254.128) |
| 09:09.18 | beaver_rrr | kaldemar: i've looked into dialstatus variable and dont get how can it help with my problem |
| 09:09.25 | beaver_rrr | i'll try to describe it better |
| 09:10.39 | beaver_rrr | local user with ext 100 cals external user, i.e. 5552222; call goes NO ANSWER. Hour/two/day passes, 5552222 calls back, and i want him to be redirected too ext.100 |
| 09:11.41 | kaldemar | what is "external" in this case? |
| 09:12.03 | beaver_rrr | external is an outbound |
| 09:12.04 | beaver_rrr | errm |
| 09:12.08 | kaldemar | as in how are you connected to outside world? |
| 09:12.20 | beaver_rrr | thru sip provider |
| 09:12.22 | kaldemar | all calls are outbound from asterisk's point of view. |
| 09:12.29 | *** join/#asterisk itsurkg (~itsurkg@202.52.236.10) |
| 09:12.41 | kaldemar | do you only have a single DID? |
| 09:12.47 | beaver_rrr | DID? |
| 09:13.04 | kaldemar | number that is dialed to reach your system |
| 09:13.15 | beaver_rrr | yes, only one. |
| 09:13.52 | *** join/#asterisk shadebob (~shadebob@41.142.254.128) |
| 09:15.01 | kaldemar | you'll have to store the caller and the called number in case of no answer somewhere, for example astdb. when you get a call in compare the caller id number against the saved value. |
| 09:15.31 | beaver_rrr | i have them saved in mysql db, good enough? |
| 09:15.47 | kaldemar | may be. |
| 09:15.53 | kaldemar | have are you saving them? |
| 09:16.03 | beaver_rrr | yes, they're saved |
| 09:16.12 | beaver_rrr | every call, with any status |
| 09:16.18 | kaldemar | sorry, s/have/how/ |
| 09:17.29 | kaldemar | oh, ok. you'd have two cases to cover with no answer. 1. app Dial reaches timeout 2. caller hangs up before answer. |
| 09:18.07 | beaver_rrr | both of them put "NO ANSWER" disposition into the base |
| 09:18.07 | kaldemar | 2 needs to be done in a hangup extension. |
| 09:18.40 | kaldemar | you don't need to touch CDR with this. |
| 09:19.22 | kaldemar | you can put the values to the DB from dialplan, and do a check on incoming calls in dialplan also. |
| 09:19.40 | beaver_rrr | i have HangUp; in the end of each entry of dialplan, if it is what you mean |
| 09:20.14 | beaver_rrr | ah, so if i have hangup with noanswer i just put this number into db |
| 09:21.18 | kaldemar | that's not what i mean. see https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions |
| 09:24.06 | beaver_rrr | ty |
| 09:24.25 | kaldemar | simplified, you could have exten => h,1,ExecIf($["${DIALSTATUS}" = "NOANSWER"]?Set(DB(NOANSWER/${CALLERID})=${DIALED_EXTENSION})) |
| 09:24.59 | kaldemar | that's using astdb and assuming that the channel variable DIALED_EXTENSION is the called number. |
| 09:26.33 | beaver_rrr | a-ha. And now what is left is just put it in my marvelous ael dialplan ._. Big thanks anyway, now i know where to look |
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| 09:54.02 | danfromuk | Hi. I'm currently in dispute with a SIP provider and need to log all the sip dialogs. Is tcpdump the only way to do this? Is there a way to log sip dialogs on a per call basis rather than one big file? |
| 10:00.32 | kaldemar | danfromuk: tcpdump is certainly not the only way. use asterisk, just enable sip debug for the peer or ip and you'll get a trace. |
| 10:03.18 | danfromuk | kaldemar: how can I write the sip trace to a file from asterisk? |
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| 10:05.27 | kaldemar | danfromuk: you can use tee or script or some other program that logs output to a file, e.g. asterisk -vvvr | tee /tmp/ast_output.log |
| 10:08.54 | danfromuk | ok, thanks. ill give it a go. |
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| 10:18.09 | ChrisInSydney | it would be nice to be able to pipe SIP traces to another file |
| 10:18.39 | ChrisInSydney | still, I suppose the source is there if you are keen |
| 10:25.27 | wdoekes2 | two words: logger.conf verbose |
| 10:27.37 | kaldemar | verbosity does not include sip debugs though. |
| 10:33.09 | wdoekes2 | in that case 'debug' |
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| 11:49.41 | xoveruk | hi |
| 11:50.02 | xoveruk | When I use a different username to my trunk name my iax2 trunk fails to authenticate, why is this? |
| 11:50.24 | xoveruk | as soon as I set the username and trunk name to the same value the two server authenticate. |
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| 12:43.58 | xoveruk | why doesnt username work when using iax2? |
| 12:46.49 | xoveruk | i deleted the username and authication still works |
| 12:46.51 | xoveruk | how? |
| 12:46.58 | xoveruk | why is it using the trunk name for authentication? |
| 12:47.10 | xoveruk | no where in the documentation does it stipulate that trunk name is used |
| 12:47.46 | [TK]D-Fender | [thisistheusernameunlessyouoverride] |
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| 13:09.10 | carrar | [manitsureisearly] |
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| 13:09.43 | carrar | [IthinkIwillmakeaLatte] |
| 13:10.30 | jaytee | [Ihatemyjobandneedavacation] |
| 13:10.43 | carrar | [IhearyaMeeToo] |
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| 13:12.28 | carrar | [makingalatte] |
| 13:12.28 | carrar | type=latte |
| 13:12.28 | carrar | allow=caffeen |
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| 13:36.54 | carrar | w00t! |
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| 13:45.36 | pithagorians | what is the default qualifyfreq ? |
| 13:47.52 | kaldemar | pithagorians: it says 60 in the sample config file. |
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| 14:20.51 | pithagorians | <kaldemar> yes, true. thx |
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| 14:35.00 | lsilva_adclick | hello everyone |
| 14:35.45 | lsilva_adclick | i'm having some issues with attended transfer on asterisk 1.8.11.0 (the latest asteriskNow iso image) |
| 14:36.12 | lsilva_adclick | is anyone aware of any bug relating to that? |
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| 14:40.07 | oyugik | Hey guys can anyone please help me with this |
| 14:40.13 | oyugik | "Starting simple switch on 'DAHDI/2-1' |
| 14:40.15 | oyugik | <PROTECTED> |
| 14:40.15 | oyugik | <PROTECTED> |
| 14:40.15 | oyugik | <PROTECTED> |
| 14:40.15 | oyugik | <PROTECTED> |
| 14:40.15 | oyugik | <PROTECTED> |
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| 14:42.22 | file | you haven't actually asked a question or described a problem so the chances of an individual helping you is not particularly high at this point in time |
| 14:43.42 | oyugik | @file that is the log output when I have an incoming call |
| 14:43.56 | file | yes, and? |
| 14:44.03 | oyugik | The problem is it doesnt show the callerid |
| 14:44.23 | file | now someone can possibly help you |
| 14:44.27 | oyugik | Executing [s@DID_trunk_2:2] ExecIf("DAHDI/2-1", "0?Set(CALLERID(all)=UNKNOWN <0000000>)") in new stack |
| 14:45.02 | oyugik | how can I go about solving the problem, |
| 14:45.37 | lsilva_adclick | My issue is A calls B; B dials feature code for attended transfer and talks to C; C hangs up; All calls terminate but A should return talking to B |
| 14:45.41 | *** part/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 14:45.52 | oyugik | is it something I need to work on from dahdi-channels or chan-dahdi.conf |
| 14:46.09 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 14:48.27 | jeffspeff | i'm having an issue with 2 phones at a remote office. if they make a call the audio works fine. if they receive a call then audio doesn't work at all. i've had both of these phones shipped to me for testing, and they work fine at our main office (which is also where the * server resides). all other phones at the remote location work fine, except for these two. I've disabled direct audio and set RTP ports. any other ideas? |
| 14:48.46 | WIMPy | oyugik: What's your question again? You got auto-ignored. |
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| 14:50.27 | oyugik | @WIMPy: my question is, when I get an inbound call. It doesnt show me the callerid |
| 14:50.51 | oyugik | the output of the log is |
| 14:50.53 | oyugik | Starting simple switch on 'DAHDI/2-1' |
| 14:50.53 | oyugik | <PROTECTED> |
| 14:50.53 | oyugik | <PROTECTED> |
| 14:50.53 | oyugik | <PROTECTED> |
| 14:50.53 | oyugik | <PROTECTED> |
| 14:50.53 | oyugik | <PROTECTED> |
| 14:51.20 | kaldemar | oyugik: nothing there is supposed to show caller id. |
| 14:52.07 | ChrisInSydney | oyugik: Had that the other week. The carrier had not turned on caller ID X/ |
| 14:52.54 | oyugik | @kaldemar when I make a call to the pstn I can see exactly which extension is calling and what number it is dialing |
| 14:53.31 | [TK]D-Fender | oyugik, first.. PASTEBIN, do not flood in here.... and show us yuor configs |
| 14:53.46 | oyugik | @ChrisInSydney: Could this be an operator issue or what? |
| 14:53.56 | [TK]D-Fender | oyugik, Show us your configs |
| 14:54.37 | oyugik | [TK]D-Fender which ones should I paste? |
| 14:54.51 | oyugik | sorry for flooding guys |
| 14:54.53 | ChrisInSydney | oyugik; SImple test if its PSTN is to plug a display handset in, if you have one |
| 14:55.03 | WIMPy | oyugik: What doesn't show it? Your log or your phone? |
| 14:55.08 | lsilva_adclick | If anyone has any idea here is the log: http://pastebin.com/JTkMbQvk |
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| 14:55.25 | ChrisInSydney | if its ISDN, then its a litle more tricky |
| 14:55.32 | WIMPy | oyugik@ |
| 14:55.37 | WIMPy | oops |
| 14:55.42 | ChrisInSydney | the job I had was ISDN, had no Caller ID turned on |
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| 14:56.03 | WIMPy | oyugik: There is nothing in your dialplan to show it. But you cann add whatever you want. See Verbose. |
| 14:56.16 | Ice_Strike | I have thousands of files i the asterisk monitor folders - I want to upload them all to other server |
| 14:56.23 | Ice_Strike | What the best way doing that |
| 14:56.38 | Ice_Strike | Im thinking rsync? |
| 14:57.11 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 14:57.22 | WIMPy | rsync has the bonus of being able to recover if it's stopped in between. Otherwise: Whatever. |
| 14:57.24 | Ice_Strike | it should not upload if the files are still recording. so I am thiking have temp name in the file name, for example: temp_123445.wav = while recordng |
| 14:57.31 | ChrisInSydney | Ice_Strike: Im thinking beer |
| 14:57.33 | Ice_Strike | When it finsih - then rename to 123445.wav |
| 14:57.41 | ravnx | Ice_Strike: ya i'd "rsync -ave ssh /local/path/ ip.ip.ip.ip:/remote/path |
| 14:58.10 | [TK]D-Fender | oyugik, Your dahdi configs clearly... |
| 14:58.12 | ChrisInSydney | rsync is simple enough |
| 14:58.39 | WIMPy | what's that ssh thing? rsync uses ssh by default. |
| 14:58.45 | Ice_Strike | I would like rsync to run every few hours - should I use cron job |
| 14:59.01 | ChrisInSydney | cron works |
| 14:59.14 | Ice_Strike | Oh when the files has been uploade - I want the files to be deleted from the main server.. |
| 14:59.28 | Ice_Strike | How to ensure files is 100% uploaded before deleting? |
| 14:59.39 | ChrisInSydney | if you want to get sophisticated, there is a thingy called iFolder by Novel (Novel who?) |
| 14:59.44 | WIMPy | rsync && rm |
| 15:00.01 | Ice_Strike | rm should not execute if rsync fail? |
| 15:00.04 | WIMPy | but there might even be an option for that. |
| 15:00.15 | WIMPy | hence the && |
| 15:01.02 | ChrisInSydney | iFolder wouldnt be any good if you want to move them. |
| 15:01.54 | WIMPy | The manual says --remove-source-files. |
| 15:03.30 | Ice_Strike | WIMPy thanks! |
| 15:12.39 | lsilva_adclick | any thoughts? evrything seems right and blind transfers work as they should |
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| 15:17.57 | leifmadsen | I'm pretty sure I know the answer is no, but is it possible to use sendrpid=pai and rpid at the same time? |
| 15:19.32 | leifmadsen | turns out when I use sendrpid=pai (for CID updates when using SLA) that it breaks callerID for my phones when placing a call |
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| 15:23.03 | WIMPy | Hmm. Who wrote in a ceratin book, that SIP is the stuff that works? |
| 15:26.19 | *** part/#asterisk oyugik (~oyugik@41.84.143.52) |
| 15:29.53 | jeffspeff | i'm having an issue with 2 phones at a remote office. if they make a call the audio works fine. if they receive a call then audio doesn't work at all. i've had both of these phones shipped to me for testing, and they work fine at our main office (which is also where the * server resides). all other phones at the remote location work fine, except for these two. I've disabled direct audio and set RTP ports. any other ideas? |
| 15:30.20 | ChannelZ | sounds like NAT issues |
| 15:30.42 | jeffspeff | ChannelZ, how would that only effect 2 out 8 phones? |
| 15:31.00 | ChannelZ | Configuration? |
| 15:31.15 | jeffspeff | all configs are the same |
| 15:31.30 | ChannelZ | And the peers on the Asterisk side? |
| 15:31.53 | jeffspeff | ? |
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| 15:34.12 | ChannelZ | Are either end behind NAT in the first place? Have you looked at a SIP debug to see what IP the phones are giving out to send their media to? |
| 15:34.23 | ChannelZ | runs out late for work |
| 15:34.44 | cusco | hi |
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| 15:39.42 | cusco | asterisk cdr insnot being properly writtin in mysql |
| 15:40.24 | cusco | it keeps on adding reccords, but empty values |
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| 16:48.10 | p3nguin | If a phone user complains that suddenly DTMF digits are being duplicated in IVRs that she calls, and rfc2833 is the dtmfmode being used, what should be checked and/or changed to try to resolve the problem? Could jitter cause duplicate DTMF to be interpreted? She said it just started yesterday, and today it seems worse. This isn't a new asterisk deployment. |
| 16:49.02 | p3nguin | I fear that Comcast is a factor in the problem. |
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| 16:57.22 | WIMPy | Do you also use rfc2833 to your itsp? |
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| 17:06.37 | p3nguin | Yes, I prefer it, so I use it everywhere. |
| 17:06.51 | p3nguin | It's odd that this problem just suddenly began. |
| 17:14.15 | ravnx | p3nguin: with rfc2833 its usually pretty decent, except with yealink phones :) I'm always battling dtmf issues. Usually though with "inband" you will see duplicated digits, with rfc2833 it isnt supposed to happen, but can, depending on where in the call it gets converted. You're only rfc2833 to the last hop that supports it, then it gets played onto the line. |
| 17:15.03 | p3nguin | But since it suddenly began, there is some factor causing the problem. |
| 17:15.21 | ravnx | I've also seen weird stuff like: [MyCustomer -> Our Core -> Paetec -> Another Paetec Customer PBX] not allowing DTMFs to work, but calling wells fargo ivr for instance works fine. |
| 17:15.32 | pabelanger | What is g722 encoded at in asterisk 1.8? 16 kHz @ 64kbits? |
| 17:15.40 | ravnx | p3nguin: you're on g711? |
| 17:15.51 | p3nguin | Yes, ulaw. |
| 17:16.06 | p3nguin | ulaw on the end points and from asterisk to the ITSP. |
| 17:17.03 | p3nguin | I doubt the ITSP made any sudden changes, so I feel like it must be something network related. |
| 17:17.12 | ravnx | usually its not jitter related. rfc2833 sends three RTP events per dtmf for redundancy |
| 17:18.40 | ravnx | could be on the other end too. does your user say its ALL dtmf attempts, or just one number they call? |
| 17:23.38 | p3nguin | I'll check with her. |
| 17:24.59 | ravnx | if you do a capture you will see the rtp dtmf events if you use wireshark to check it out. but might not help usually its fine at the asterisk box |
| 17:25.26 | p3nguin | The phones are SPA-942. Does the DTMF Playback Length setting pertain to DTMF being sent by the phone into the call, or only the DTMF that is received from the call into the phone? |
| 17:26.09 | ravnx | received and played |
| 17:26.15 | p3nguin | (I don't know how to interpret the word "playback" in this context.) |
| 17:26.47 | ravnx | its on the receiving end, since it doesnt play dtmf to the call |
| 17:30.48 | ravnx | on that SPA, it should be set to AVT for rfc2833 and not auto. (even though auto should work fine) |
| 17:32.36 | p3nguin | She said it's more than just one place. At first I asked if it was only one, and she said two. Then I asked if she tried other numbers besides those two, and she said all the places she's called all day today. |
| 17:33.11 | p3nguin | She calls financial institutions and their systems usually ask for account number and SSNs... |
| 17:33.30 | p3nguin | So I can see how it's a problem. |
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| 17:35.06 | ravnx | yeah ive got a customer that does insurance and another that does medical billing, they always complain when theres a problem. |
| 17:35.07 | p3nguin | The phones are set to auto and asterisk is set to rfc2833. I don't expect it to make any difference, but I will change the phones to AVT anyway. |
| 17:35.18 | ravnx | i doubt it will make any difference |
| 17:35.22 | ravnx | just a long shot |
| 17:35.22 | p3nguin | It shouldn't. |
| 17:35.35 | *** join/#asterisk greenwolf (~frozensol@cpe-76-180-217-47.buffalo.res.rr.com) |
| 17:35.44 | p3nguin | But since I like uniformity, I'll change them all anyway. |
| 17:36.18 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 17:36.42 | ravnx | might call up the ITSP just to check, if they moved termination traffic to another provider it could happen. |
| 17:41.12 | ravnx | i'd bet its at the ITSP or at their provider |
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| 18:33.21 | *** join/#asterisk mbrevda (~users.550@gateway/web/irccloud.com/x-mpvhkizwjggxfgmc) |
| 18:33.31 | mbrevda | does anyone have asterisk running on os x? |
| 18:39.24 | *** join/#asterisk jshriver (~jshriver@136.sub-174-231-2.myvzw.com) |
| 18:39.28 | jshriver | Greetings everyone |
| 18:39.52 | jshriver | anyone have an example of what a extensions.conf dialplan should look like for roll over with voip extensions? |
| 18:39.58 | jshriver | or if it needs configured anywhere else |
| 18:45.08 | [TK]D-Fender | jshriver, Your definition is too vague. |
| 18:45.20 | [TK]D-Fender | jshriver, Show us what you've got an where you'd like to do more... |
| 18:46.51 | jshriver | http://pastebin.com/Ca8F9CHB |
| 18:47.13 | jshriver | so if 300 and 301 don't answer try another set of extensions, and so on |
| 18:47.21 | [TK]D-Fender | jshriver, So what is there to "roll over" in there? |
| 18:47.32 | [TK]D-Fender | just shove another dial after it |
| 18:47.40 | jshriver | Didn't know if I could just add another Dial line and list the next wave of phones |
| 18:47.45 | jshriver | ah ok. |
| 18:47.55 | [TK]D-Fender | You just shoved Voicemail there. You can do whatever yuo want |
| 18:48.04 | [TK]D-Fender | So more dials it is.... |
| 18:48.06 | jshriver | didnt know if it was that simple. Or if there had to be some kind of logic |
| 18:48.09 | jshriver | thank you very much |
| 18:48.16 | [TK]D-Fender | Nope, entirely that simple. |
| 18:48.30 | jshriver | by chance do you play L4D2? A guy I play with has the same handle there. |
| 18:48.36 | [TK]D-Fender | Dialplan is about as linear as it looks. |
| 18:48.51 | jshriver | bbiab testing |
| 18:48.53 | [TK]D-Fender | jshriver, I've had this nick for at least a decade now... |
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| 21:43.13 | gusto | hi ppl |
| 21:43.16 | gusto | what's up? |
| 21:45.03 | dtascom | hi there |
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| 21:56.07 | Micc | does asterisk allow for multiple odbc databases in case one fails? |
| 21:56.24 | Micc | I know you can configure multiple databases, but is there a way to setup multiple servers for a single database? |
| 22:00.41 | *** join/#asterisk michael-i (~anonymous@204.11.231.178.static.etheric.net) |
| 22:10.02 | ravnx | Micc: mysql-ndb |
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| 22:20.09 | *** join/#asterisk em_pleh (46be07fd@gateway/web/freenode/ip.70.190.7.253) |
| 22:20.17 | em_pleh | Hello friends! |
| 22:20.58 | Chainsaw | Hi! |
| 22:21.08 | michael-i | I want to execute hangup logic after every call in my system. Is there a best-practice on where to put those h extensions? Dialplan origination or termination? Some global h extension w/goto's? |
| 22:21.30 | em_pleh | :) |
| 22:23.34 | gusto | eh |
| 22:23.44 | [TK]D-Fender | michael-i: there is no "global". Design your dialplan accordingly |
| 22:23.48 | gusto | does someone already use g722? |
| 22:24.08 | em_pleh | I recently upgraded to asterisk 1.8 and i have run into a wall. My inbound calls work for a good five minutes and then i start getting dead air. I have copied my log of a incoming call that goes no where http://pastebin.com/1EWjfVrg and my config is as fallows http://pastebin.com/3ikXtW6T . the settings where all the same except i changed the insecure=very to insecure=port,invite. Any insite would be appreciated |
| 22:24.10 | [TK]D-Fender | gusto: No. we've had it for years and we've been waiting for that one magical person to lead us! |
| 22:24.39 | em_pleh | #Also i am not sure if it matters or not but my sip provider is Inphonex |
| 22:25.11 | michael-i | [TK]D-Fender: I know. Just curious if there was a best practice on emulating that. |
| 22:25.50 | gusto | [TK]D-Fender: but there are now gigaset's that can do that |
| 22:26.01 | michael-i | I'm going to make a [logic] context and just h->goto(logic,1) in each phone/provider's context |
| 22:26.11 | [TK]D-Fender | michael-i: Put "h" in every context your calls may reach |
| 22:26.32 | file | em_pleh, there is no incoming call in that log. |
| 22:27.11 | em_pleh | file: line 24 |
| 22:27.26 | em_pleh | thats what gets out as the call is initiated |
| 22:27.32 | file | that's not an incoming call, that's a subscription for message waiting indicator from a Mitel phone |
| 22:27.55 | michael-i | [TK]D-Fender: seems dirty, but it's as I was expecting… I know, I know "patches welcome" |
| 22:28.42 | em_pleh | file: thats weird everytime i try to make a incoming call that comes up |
| 22:28.47 | em_pleh | is there any connection? |
| 22:29.01 | [TK]D-Fender | michael-i: Can't say I see a need for apatch |
| 22:29.01 | file | em_pleh, it's possible but doubtful ... |
| 22:29.30 | file | em_pleh, more likely a NAT mapping is expiring causing the incoming INVITE from your provider to be dropped |
| 22:30.03 | em_pleh | file: ahh could that be why when it does work and you hangup the call it still continues to ring? |
| 22:30.24 | file | em_pleh, it's possible but without a log of that scenario it is impossible to say for certain |
| 22:30.45 | file | em_pleh, you can attempt to create a peer entry with the same host that you register to with qualify set to yes - this may be enough to force the NAT mapping to stay open from your side |
| 22:31.49 | em_pleh | file: i did a sip reload so that incoming will work, this is the log http://pastebin.com/Jy7WXTrW |
| 22:33.01 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-fkioysdwfykgwbfo) |
| 22:33.02 | michael-i | [TK]D-Fender: just a global callback to "always execute this logic on a hangup" then progress to the local h logic. Definition of "nitpick.." |
| 22:33.39 | [TK]D-Fender | michael-i: Or as we prefer to call it "the very reason the h extension already exists and works exactly as advertised" |
| 22:33.57 | file | em_pleh, it looks as though something is interfering with the traffic outside of Asterisk - be it the NAT, or a SIP aware extension - the messages Asterisk is sending are not making it to the ITSP |
| 22:34.18 | em_pleh | file: will qualify help this? |
| 22:34.41 | file | qualify will potentially keep the NAT mapping open but if something is interfering with the traffic outside of that... then no, it's not magic |
| 22:35.05 | em_pleh | hummm... what would my next step be into troubleshooting this issue |
| 22:35.28 | file | turn qualify on and see what happens. |
| 22:36.17 | file | opening up and forwarding the ports on your firewall would also do it |
| 22:37.37 | em_pleh | I have done that already |
| 22:38.11 | file | if the traffic never shows up in Asterisk and the assumption that your ITSP is sending it to you is true... then something is stopping it ... |
| 22:38.24 | em_pleh | ok let me look things up |
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