| 00:16.59 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 00:31.32 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 00:31.59 | ageis | anyone know syntax for sendmailmp3 if I want to use it outside of Asterisk's mailcmd? |
| 00:34.29 | Cartes | ageis: install cli-lame encoder, and put it into the script before sending out the email |
| 00:35.15 | ageis | Cartes: I have all the script's requirements already... I'm asking how to use it on the command line. For instance, sendmailmp3 to@this.email < msg0.wav |
| 00:35.50 | ageis | I'm trying to adapt it for something else |
| 00:35.56 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 00:39.34 | ageis | Cartes: any idea ?? |
| 00:41.04 | Cartes | ooo |
| 00:41.29 | *** join/#asterisk luckman212_ (~luckman21@unaffiliated/luckman212) |
| 00:41.45 | Cartes | hold on |
| 00:42.24 | Cartes | ageis: is the "sendmailmp3" a bash script? |
| 00:42.30 | Cartes | ageis: did you read this: |
| 00:42.32 | Cartes | http://bernaerts.dyndns.org/linux/179-asterisk-voicemail-mp3 |
| 00:42.53 | ageis | Cartes: yes, from pbxinaflash.com |
| 00:43.26 | ageis | Cartes: yes I did read that, but this covers usage as Asterisk's mailcmd for voicemail, NOT syntax |
| 00:43.54 | ageis | Given an e-mail address and an audio file, I am wondering how I would structure a command to use this script outside of Asterisk, do you understand? |
| 00:44.08 | Cartes | aah ok... |
| 00:44.14 | ageis | perhaps someone who knows sendmail could help |
| 00:44.15 | Cartes | use "arguments" |
| 00:44.29 | ageis | hrm ok |
| 00:44.34 | Cartes | do you know how to pass arguments to Bash Scripts? |
| 00:44.45 | ageis | yes I do... silly question |
| 00:44.50 | Cartes | sorry |
| 00:45.24 | ageis | I'm a certified and full-time sysadmin.. I am confused by that approach because this script calls sendmail -t and doesn't pass any other arguments to it, just a stream that's supposed to include the message. |
| 00:45.32 | Katty | helllllllllllllllllllloooooooo asterisk. |
| 00:45.35 | ageis | So I assume I have to construct an input stream with the message, audio file and recipient in it. |
| 00:46.03 | Cartes | ageis: oh... then you actually are "over thinking" it... :-) |
| 00:46.46 | ageis | if I knew something about how asterisk calls mailcmd and what it passes to it, then it would help.. |
| 00:47.28 | Cartes | I actually was reading up on this yesterday |
| 00:47.34 | Cartes | when I was configuring our voicemail |
| 00:48.10 | Cartes | what are you trying to achieve? |
| 00:48.22 | Cartes | I think I misunderstood your question |
| 00:48.40 | ageis | Cartes: A simple one-liner using this script that will send my already-recorded WAV to an e-mail address of my choice. |
| 00:49.21 | Cartes | ok. simple enough... why do you want to do it externally? |
| 00:49.48 | ageis | Cartes: I have an implementation that uses the Record() function.. because I am not using Voicemail and it won't be triggered that way. |
| 00:50.07 | Cartes | ok |
| 00:50.31 | ageis | but who cares about the why... |
| 00:50.42 | Cartes | so, you want to trigger sending the mail from the dialplan |
| 00:50.47 | Cartes | after the recording is complete |
| 00:51.08 | ageis | yep and I'm using a System call to do that... |
| 00:51.23 | Cartes | so, if you have a script that takes in params, and pass on the email + wav recording... that should do it, not? |
| 00:52.33 | Cartes | you could use the "Shell" dialplan funciton to trigger the external script |
| 00:52.34 | ageis | Cartes: I need an example dude.. I need the syntax that would do that... do I just put my e-mail address and the recording on the other side of > ??? |
| 00:52.42 | ageis | not relevant |
| 00:52.57 | Cartes | http://www.voip-info.org/wiki/view/Asterisk+func+shell |
| 00:53.22 | ageis | not relevant at all. I'm trying to pass the recipient parameter and the recording into this script. How is it structured, how does it know what's what |
| 00:53.26 | Cartes | ${SHELL(mymailerscript.sh myself@mydomain.com blahblah.wav):0:-1} |
| 00:53.34 | ageis | ok |
| 00:53.45 | Cartes | that's from the dialplan |
| 00:54.26 | ageis | I need the code for this mymailerscript.sh |
| 00:54.30 | ageis | that doesn't tell me anything |
| 00:54.34 | ageis | about what I need to know |
| 00:54.40 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 00:54.53 | ageis | My question is about how the input stream for sendmail, including an attachment and recipient needs to be structured |
| 00:54.59 | ageis | very specific |
| 00:55.58 | Cartes | I think FreePBX has a nice implementation on that |
| 00:56.05 | Cartes | let me see if i can dig that out |
| 00:56.16 | ageis | ty |
| 00:58.17 | Cartes | you've got sendmail working, right? or do you need help on that too? |
| 00:58.49 | *** part/#asterisk ghost75 (~trechber@dslb-188-105-016-160.pools.arcor-ip.net) |
| 01:00.11 | ageis | nope |
| 01:00.15 | ageis | Cartes: I got it working |
| 01:00.24 | Cartes | great... |
| 01:00.25 | ageis | erm |
| 01:00.33 | ageis | nvm the mail came up all corrupted |
| 01:00.39 | ageis | yes I have sendmail working |
| 01:01.51 | Cartes | so a simple... mail -s "Recorded Message" -c myself@mydomain.com < myfile.wav |
| 01:01.58 | Cartes | does that work from the command line? |
| 01:02.46 | ageis | yes that should do the trick |
| 01:02.46 | ageis | or |
| 01:03.20 | ageis | uuencode myfile.wav | mail -s "Recorded Message" myself@domain.com |
| 01:03.52 | Cartes | if you run either of them from the CLI, manually, does the email get routed properly? |
| 01:04.32 | ageis | Cartes: yes, why? my problem is the file not being attached properly.. the MIME content type... the audio data is pasted into the e-mail body |
| 01:04.38 | ageis | Cartes: I think I got this now... thanks. |
| 01:05.06 | Cartes | no problem... although i don't think I was of much help |
| 01:05.07 | Cartes | :-) |
| 01:06.04 | ageis | hehe |
| 01:06.34 | Cartes | what fixed it? |
| 01:06.45 | ageis | aha.. that uuencode command works great |
| 01:07.12 | ageis | im ditching this sendmailmp3 script.. its all unnecessary, cutting the stream into parts, encoding base64 etc |
| 01:07.15 | ageis | when uuencode does the strick |
| 01:07.20 | ageis | *trick |
| 01:07.32 | Cartes | :-) |
| 01:08.19 | *** join/#asterisk willryder (~tedryder@nc-184-3-97-125.dhcp.embarqhsd.net) |
| 01:09.20 | coppice | if you are happy with the wave file being the body of the message, the simple uuencode approach is OK. most people want to send a text note, with the wave file attached. That requires the sendmail scripts |
| 01:10.39 | Cartes | has anyone here worked with "Follow Me" over SIP & TDM? |
| 01:11.38 | joako | Cartes, Not the actual ¨follow me¨ application, but same concept yes |
| 01:11.41 | ageis | Cartes: last problem... I need to send the Ctrl+D to the input of the mail command, or specify a message, do u know the argument/how? |
| 01:12.17 | ageis | Cartes: nvm.. I will append it to the stream |
| 01:12.22 | Cartes | ageis: no, not sure... I actually wanted to go all scripty... |
| 01:13.13 | Cartes | joako: how do I tell another asterisk over SIP that the call is a forwarded call, and provide all the right phone numbers: Original Caller, Forwarded Caller and Destination? |
| 01:13.53 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:14.48 | joako | Cartes, what do you mean by ¨forwarded caller¨ |
| 01:15.07 | Cartes | forwarded meaning, when a local extension is unavailable... |
| 01:15.20 | Cartes | that local extension will "forward" the call to an outside cell phone |
| 01:15.28 | Cartes | so, this local extension is the "forwarded caller" |
| 01:16.33 | joako | Cartes, Then how does that differ from ¨Destination?¨ |
| 01:16.54 | Cartes | oh... because the "Cell Phone" is the "new Destination" |
| 01:16.55 | Cartes | :-) |
| 01:17.07 | Cartes | let me put the call flow for easier understanding |
| 01:17.19 | joako | Right but I am confused why the cell phone needs to know its own number like that |
| 01:17.38 | Cartes | caller -> [calls] > local-number -> [call fails] > forward-to-cell |
| 01:17.47 | Cartes | this is the call flow on my PBX |
| 01:18.54 | Cartes | to complete the call on the "cell", i need to send the call to another Asterisk, which acts as the TDM / SS7 Media Gateway |
| 01:20.35 | joako | So you would use the sip header ¨Diversion: ¨ to send ¨local-number.¨ CallerID stays the same that´s how you get the caller info, Diversion has your forwarded caller and destination would be the Destination in your Dial() string |
| 01:20.54 | joako | And if everything is working correctly your cell phone should indicate the call is forwarded |
| 01:22.11 | joako | Just like: http://i.imgur.com/SuAR3Ot.jpg |
| 01:22.36 | Cartes | wow... :-) |
| 01:23.11 | joako | In the dialplan you could use something such as: SipAddHeader(Diversion: <sip:+${EXTEN}@my.realm.com>) |
| 01:24.25 | joako | Your PRI or SS7 gateway should hopefully translate this to the proper field for RDNIS |
| 01:24.54 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:24.56 | Cartes | does the recent libss7 version support RDNIS and OCN? |
| 01:25.28 | joako | I have absolutely no experience with that. But it does work with DAHDI PRI |
| 01:25.50 | Cartes | ok... i was reading elsewhere, that Atilla's libss7 would be needed |
| 01:25.59 | Cartes | and that dates back to asterisk v 1.6 |
| 01:26.12 | Cartes | thx Joako |
| 01:26.18 | Cartes | at least part of the problem is solved! |
| 01:27.51 | joako | Just one more thing to note: not all cell phone carriers support this... but it should usually work on GSM/UMTS and last time I checked iPhone didn´t display the forwarded flag |
| 01:35.56 | *** join/#asterisk iq (~iq@107-196-241-2.lightspeed.rcsntx.sbcglobal.net) |
| 01:37.19 | Cartes | joako: yeah, i'm not worred about the forwarded flag right now... what's problematic for me, the telco who we send the call over SS7 does not accept calls if we do not put the "right" caller |
| 01:37.20 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:37.39 | *** join/#asterisk Bradada (~Bradada@59-120-137-34.HINET-IP.hinet.net) |
| 01:37.46 | Cartes | joako: in cases like this, they need the right params on the SS7 RDNIS and OCN fields |
| 01:39.08 | Cartes | joako: for now, if the caller is from within our PABX, then the forwarded call goes through... as the "caller" in this case has the right format (digits at the front) |
| 01:46.46 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 01:47.15 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:53.43 | joako | Cartes, If you´re forwarding calls then you really need to get the carrier to open up the line and let anything through |
| 01:56.59 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 02:22.45 | *** join/#asterisk DND (5c60fb5d@gateway/web/freenode/ip.92.96.251.93) |
| 02:23.53 | DND | guys need help i did dahdi_scan and i seee that i have irq=0 and i get Primary D-Channel on span 1 up Primary D-Channel on span 1 down |
| 02:24.00 | DND | how can i resolv e this? |
| 02:28.55 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 02:30.14 | ageis | whats a good cheap trunking provider for personal usage? |
| 02:40.15 | ChannelZ | How much usage? |
| 02:40.24 | phix | ageis: pennytel.com |
| 02:40.31 | phix | for AU :) |
| 02:40.36 | ChannelZ | and it depends on what your threshold is for "cheap" |
| 02:43.11 | phix | $10 for a DID every 2 years, $0.08 AUD untimed no flag fall per national call and to ~20 countries inc US, UK, DK, etc.. (usually land line only although untimed calls to singapore and a few other countries with an awesome phone network) |
| 02:44.03 | phix | up to 4 concurrent channels if paying via credit card |
| 02:44.23 | phix | pre-paid with auto topup to a set amount that you define per month |
| 02:52.02 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 02:53.56 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 03:01.36 | *** join/#asterisk Bradada (~Bradada@220-135-49-159.HINET-IP.hinet.net) |
| 03:05.30 | *** join/#asterisk jkroon (~jkroon@kerberos.uls.co.za) |
| 03:18.19 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 03:18.48 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 03:21.03 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 03:26.27 | *** join/#asterisk elmargol (~elmargol@host44-23-dynamic.20-79-r.retail.telecomitalia.it) |
| 03:33.11 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 03:37.34 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 03:39.46 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 03:50.39 | *** join/#asterisk maslowbeer (~maslowbee@pool-71-177-17-63.lsanca.dsl-w.verizon.net) |
| 03:50.48 | *** join/#asterisk quenenni (~quenenni@unaffiliated/quenenni) |
| 03:51.01 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 03:51.39 | *** part/#asterisk quenenni (~quenenni@unaffiliated/quenenni) |
| 04:00.49 | *** join/#asterisk FireAndIce (~FireAndIc@175.100.158.241) |
| 04:19.27 | *** join/#asterisk jkroon (~jkroon@kerberos.uls.co.za) |
| 04:33.49 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 05:14.47 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:19.06 | ageis | in my dialplan, how can I get commands to continue to run after a non-zero hangup? |
| 05:20.06 | [TK]D-Fender | ageis: "h" Asterisk Standard Extension |
| 05:20.13 | [TK]D-Fender | ageis: Read up on your dialplan basics... |
| 05:20.23 | ageis | yeah, got it |
| 05:20.31 | ageis | so, I can put an exten => h after an exten => s ? |
| 05:20.58 | igcewieling | ageis: the order does not matter |
| 05:24.23 | ageis | it's not working |
| 05:25.25 | ageis | here's my dialplan: http://pastebin.com/wzQs3eUX |
| 05:26.15 | ageis | erm |
| 05:26.19 | ageis | need to put 1 in there |
| 05:26.21 | ageis | priority :D |
| 05:27.22 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.107) |
| 05:27.38 | [TK]D-Fender | Somewhat important.... |
| 05:28.09 | ageis | still not working... |
| 05:31.14 | kaldemar | show the whole call flow. how does the call end up in main? |
| 05:31.17 | ageis | ok, got it... only works with one, not two lines |
| 05:31.26 | ageis | kaldemar: not relevant, but thanks |
| 05:32.48 | kaldemar | oh it is relevant even if you don't know it. |
| 05:33.49 | ageis | :) |
| 05:44.39 | igcewieling | pastebin the cli output |
| 05:46.41 | *** join/#asterisk timahvo1 (~rogue@mail.sbakenyaltd.com) |
| 05:47.17 | igcewieling | mutters something about horses and water and goes back to work |
| 05:47.55 | [TK]D-Fender | igcewieling: That's right. You lead them to it, but the SPCA won't let you hold their head's under ;) |
| 05:51.03 | igcewieling | Doctor: Your tests show that you have diabetes. ageis: You're wrong. |
| 05:59.39 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 05:59.40 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 06:10.06 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 06:14.24 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 06:15.08 | *** join/#asterisk bulkorok (~chatzilla@85.183.36.36) |
| 06:16.28 | bulkorok | hi |
| 06:20.01 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 06:20.02 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 06:21.04 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 06:27.19 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 06:34.18 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 06:46.50 | *** join/#asterisk andy09usa (~Andrey@unaffiliated/andy09usa) |
| 06:48.19 | phix | hai bulkorok |
| 07:10.42 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.80) |
| 07:17.28 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 07:18.45 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 07:18.45 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-okbvbqthabzzppwc) |
| 07:22.14 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 07:24.08 | *** join/#asterisk sjs205 (~sjs205@host81-151-44-189.range81-151.btcentralplus.com) |
| 07:33.08 | *** join/#asterisk Kraln (~kraln@69.169.90.240) |
| 07:34.44 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 07:38.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.80) |
| 07:39.37 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.80) |
| 07:44.21 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:53.34 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 08:14.30 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 08:16.29 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 08:21.13 | Rhomber | Is there anyway to get asterisk to log more useful messages than "chan_sip.c:25081 handle_request_invite: Sending fake auth rejection for device 50<sip:50@202.45.126.149>;tag=ee2e56ae" |
| 08:21.14 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 08:21.27 | Rhomber | that IP address it the IP of the server |
| 08:21.52 | Rhomber | i get like 50 of these a day, but nothing is logged that I can do anything about - so the log is completely meaningless |
| 08:22.07 | *** join/#asterisk micols (~t@shell1.rlogin.dk) |
| 08:22.19 | WIMPy | It's the URL supplied by the remote. |
| 08:23.04 | Rhomber | yeah, useless. it should log the remote IP |
| 08:23.14 | Rhomber | so fail2ban can ban something |
| 08:23.31 | Rhomber | atm, i haven't bothered to even install fail2ban as it would be a waste of time |
| 08:24.39 | Rhomber | just a bit annoyed after wading through forum posts from 2011 to try and find a solution to a problem occuring in 2013 :) |
| 08:24.59 | Rhomber | did everyone give up on security last year or something? |
| 08:25.34 | Rhomber | (referring to: http://forums.asterisk.org/viewtopic.php?f=13&t=78988) |
| 08:25.54 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 08:26.00 | WIMPy | is in the group who think that something like fail3ban is not a good idea anyway. |
| 08:26.24 | WIMPy | But I agree that the logs should make sense. |
| 08:26.34 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 08:27.13 | Rhomber | "This issue is already fixed in Asterisk 10+ by including the appropriate message in the Security Event Framework." i just read |
| 08:27.34 | Rhomber | i don't mind .. i've never used fail2ban.. care to point me in the right direction? |
| 08:28.36 | kaldemar | Rhomber: http://pastebin.com/xfJZi8js |
| 08:29.41 | Rhomber | im running asterisk 11 i should add |
| 08:29.43 | Rhomber | *looks* |
| 08:29.53 | kaldemar | that patch was for 11.2.0. |
| 08:30.40 | Rhomber | [root@staff ~]# asterisk -V |
| 08:30.41 | Rhomber | Asterisk 11.2.1 |
| 08:31.11 | Rhomber | and ok, thanks.. is there a better way to do security in asterisk then? since no one has done anything since 2011? |
| 08:31.16 | Rhomber | or am i missing something? |
| 08:33.17 | WIMPy | It's just the usual issue. Be secure or be online. |
| 08:34.01 | Rhomber | lol |
| 08:34.27 | Rhomber | well i want to use it via 3G.. and i think a tunnel/vpn on the phone might create too much overhead |
| 08:34.36 | Rhomber | so not a lot of choice |
| 08:34.43 | WIMPy | It's quite serious. |
| 08:34.53 | Rhomber | what's that? |
| 08:35.07 | WIMPy | The be online or be secure thing. |
| 08:36.03 | WIMPy | And I don't see why a modern phone should have issues with a vpn. |
| 08:36.31 | Rhomber | it's not the phone .. it's the network |
| 08:36.38 | Rhomber | i live in Australia |
| 08:36.45 | Rhomber | so we have a lot of cell tower congestion |
| 08:37.15 | WIMPy | Ok, if that's bad, there may be an issue. But if it's that bad you might not want to do VOIP there anyway. |
| 08:37.56 | Rhomber | it falls back to DID, but i just wanted to give it as much of the bandwidth as was available |
| 08:38.09 | Rhomber | i don't want to be talking to a customer and have it sound shitty because of VPN overheads |
| 08:38.26 | Rhomber | also, not really keen to setup a VPN on my phone incase it gets stolen |
| 08:38.51 | kaldemar | 11.3 does not even seem to mention the fake auth rejection in the logged messages. |
| 08:38.53 | Rhomber | would probably then need to setup a second VPN and additional ip's/routing.. pain in the ass |
| 08:39.09 | Rhomber | nope? what does it mention? |
| 08:40.21 | kaldemar | just ast_log(LOG_NOTICE, "Failed to authenticate device %s for SUBSCRIBE\n", sip_get_header(req, "From")); |
| 08:40.38 | Rhomber | patch compiled and installed, we'll see if it works soon i guess.. the failed requests occur quite regularly |
| 08:40.39 | kaldemar | or ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", sip_get_header(req, "From")); |
| 08:40.53 | Rhomber | ah fair enough |
| 08:40.59 | Rhomber | still not very useful i see |
| 08:41.30 | Rhomber | I suspect i'm meant to use "This issue is already fixed in Asterisk 10+ by including the appropriate message in the Security Event Framework." |
| 08:41.34 | Rhomber | what ever that is |
| 08:41.46 | Rhomber | mentioned: https://issues.asterisk.org/jira/browse/ASTERISK-20506 |
| 08:41.47 | LieutPants | [ASTERISK-20506] [Status: Closed] With alwaysauthreject=yes AND allowguest=no Asterisk fails to report Attacker's IP Address - https://issues.asterisk.org/jira/browse/ASTERISK-20506 |
| 08:42.00 | kaldemar | applying your own patches is a PITA. |
| 08:42.22 | WIMPy | thinks that logs should provide the information neccessary to find out what happened earlier on. |
| 08:42.47 | Rhomber | yeah, yours seems to work.. even if it only provides me some visual aids while i work on the better solution is good.. so i can see if it's a spoofed ip or what |
| 08:42.59 | Rhomber | agreed |
| 08:43.10 | Rhomber | it sure wouldn't hurt to include where they came from |
| 08:43.18 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Framework |
| 08:44.20 | Rhomber | thanks, will have a read after some dumplings :) |
| 08:45.30 | WIMPy | I don't expect logs to be used for (sort of) realtime action. |
| 08:47.28 | Rhomber | nope, they shouldn't be |
| 08:47.38 | Rhomber | but they should be able to be used to discover what happened |
| 08:48.06 | *** join/#asterisk chris-NB (~chris@fw.commpany.at) |
| 08:50.24 | Rhomber | chan_sip.c:25083 handle_request_invite: Sending fake auth rejection to 37.8.126.62 for device 88<sip:88@202.45.126.149>;tag=6a040750 |
| 08:50.29 | Rhomber | didn't take long to show up |
| 08:50.33 | Rhomber | looks a lot more useful |
| 08:51.20 | *** join/#asterisk micols (~t@shell1.rlogin.dk) |
| 08:52.09 | Rhomber | I just added "security_log => security" to my logger.conf, we'll see what that produces |
| 08:53.37 | Rhomber | [Apr 11 18:52:43] SECURITY[5549] res_security_log.c: SecurityEvent="InvalidAccountID",EventTV="1365670363-132700",Severity="Error",Service="SIP",EventVersion="1",AccountID="0011972599244732",SessionID="0xb96e8d8",LocalAddress="IPV4/UDP/202.45.126.149/5060",RemoteAddress="IPV4/UDP/37.8.126.62/26654" |
| 08:53.40 | Rhomber | didn't take long |
| 08:54.06 | *** join/#asterisk dirk_ (~dirk@208.74.182.4.static.etheric.net) |
| 08:54.25 | WIMPy | Wow. |
| 08:54.40 | dirk_ | Wow? |
| 08:54.56 | WIMPy | I only get a bunch of them every few days. |
| 08:55.17 | dirk_ | missing context.. sorry |
| 08:55.20 | dirk_ | 2218 dial_exec_full: Unable to create channel of type 'Motif' (cause 66 - Channel not implemented) |
| 08:55.21 | *** join/#asterisk bombev (~User@PPPoE-Static-40-132.UnicsBG.Net) |
| 08:55.43 | dirk_ | anybody know what that means? am setting up and asterisk capable of talking to google chat/voice |
| 08:55.53 | WIMPy | Was that a question or the answer to an implied question? |
| 08:55.54 | dirk_ | googling instructions and following them |
| 08:55.57 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 08:57.35 | Rhomber | WIMPy: well, i'm not advertising.. lol. not sure why |
| 08:58.33 | Rhomber | not entirely sure how the security event framework helps me out of the box though |
| 08:58.38 | Rhomber | will investigate later |
| 09:00.16 | bombev | hui |
| 09:00.18 | bombev | hi |
| 09:00.33 | bombev | what is the reason to get: |
| 09:00.42 | bombev | [2013-04-11 10:57:51] WARNING[3564] chan_sip.c: username mismatch, have <Fayn_out>, digest has <> |
| 09:00.42 | bombev | <PROTECTED> |
| 09:00.42 | bombev | [2013-04-11 10:57:51] NOTICE[3564] chan_sip.c: Failed to authenticate device <sip:00359885092216@sip.fayn.cz>;tag=02CEFFD7-51667B0500003A40-61D2A700 |
| 09:08.47 | dirk_ | : 2218 dial_exec_full: Unable to create channel of type 'Motif' (cause 66 - Channel not implemented) |
| 09:08.51 | dirk_ | ? |
| 09:09.04 | *** join/#asterisk micols (~t@shell1.rlogin.dk) |
| 09:09.08 | dirk_ | do I not have the chan_motif driver? |
| 09:09.14 | dirk_ | or is this something else? |
| 09:09.35 | WIMPy | At least it's not loaded. |
| 09:09.48 | dirk_ | where would it be on ubuntu? |
| 09:15.18 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:15.28 | dirk_ | test |
| 09:25.45 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:25.47 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:29.34 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 09:32.55 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 09:35.28 | asteriskATmarmuD | hi guys! any hints on how to cope with "UDPTL initialization failure" while sending and receiving fax? I want to use T.38 with asterisk 10. |
| 09:39.17 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 09:55.19 | Rhomber | it kinda sucks that the new asterisk security framework is pretty useless... nothing uses it yet.. lol |
| 09:55.56 | Rhomber | or at least, there is nothing on google on modules that hook in to ban.. or log parsers that parse /var/log/asterisk/security |
| 09:58.47 | wdoekes | Rhomber: fail2ban? |
| 09:59.04 | WIMPy | Security is individual. |
| 09:59.54 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 10:01.11 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 10:07.10 | *** join/#asterisk bjhaid (~bjhaid@197.152.49.206) |
| 10:08.19 | *** join/#asterisk MalMen (~MalMen@bl6-156-183.dsl.telepac.pt) |
| 10:08.21 | MalMen | hello |
| 10:08.39 | MalMen | anyone here know how to configure sapo voip sip to asterisk ? |
| 10:08.55 | bjhaid | hi all, I would like to know if there is anyone who has a ussd integration to Asterisk via a datacard something similar to this: http://code.google.com/p/datacard/wiki/Manual cuz that hasnt been updated in over 2 years |
| 10:15.21 | WIMPy | Look for chan_dongle. |
| 10:16.03 | *** join/#asterisk n3hxs (~ed@pool-108-16-94-10.phlapa.fios.verizon.net) |
| 10:19.32 | andross | what would be cool is if android had a sip package |
| 10:19.43 | andross | which turned your cell into a sip device |
| 10:19.51 | andross | that you could register and send calls to and from |
| 10:20.06 | andross | either over 3g/4g or wifi or usb |
| 10:20.12 | WIMPy | What's wrong with having many to choose from? |
| 10:20.59 | andross | the more choices and flexibility the better |
| 10:22.07 | Rhomber | wdoekes: fail2ban sure. but i can't find an example to handle the security log and the docs are useless as they don't list the actual event names just english words.. like.. "Invalid Account Id" when it's InvalidAccountID ... but others aren't quite as easily translated |
| 10:22.22 | Rhomber | so, cbf really trying to devise my own fail2ban config for it |
| 10:23.40 | kaldemar | Rhomber: just because the framework isn't widely used YET doesn't make it useless. |
| 10:23.55 | Rhomber | sure, i get that. but useless to me right now |
| 10:24.13 | Rhomber | and ideally it would be a module within asterisk that uses it rather than a log parser |
| 10:24.24 | Rhomber | and im sure there will be such a module at some point |
| 10:25.08 | Rhomber | but to get people to use it, they should improve the docs to include the actual event names and other relevant syntaxes |
| 10:26.32 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 10:29.23 | kaldemar | Rhomber: did you look at the wiki pages? the event names are listed there. |
| 10:31.49 | bjhaid | @WIMPy was chan_dongle for me |
| 10:31.50 | bjhaid | ?? |
| 10:32.53 | WIMPy | bjhaid: yes |
| 10:33.23 | Rhomber | kaldemar: sure I did, but not the right section :) |
| 10:33.36 | Rhomber | thanks, might have a look at seeing if i can get it to work |
| 10:34.04 | bjhaid | @WIMPy: thanks alot, any where to see documentation/sample config for using chan_dongle ?? |
| 10:34.15 | Rhomber | if i can get it to work, posting the config somewhere might help someone else too lol |
| 10:34.21 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:35.36 | WIMPy | bjhaid: Doesn't it have a sample config? |
| 10:35.58 | bjhaid | @WIMPy: thanks, wanted to read ahead before I tried an installation |
| 10:37.00 | *** join/#asterisk hirnukuono (U2FsdGVkX1@goaar.finnishplatoon.org) |
| 10:37.26 | hirnukuono | hullo, i got a nifty ast 1.8 + queue + transfer problem, can i ask for help? |
| 10:37.33 | kaldemar | ~ask |
| 10:37.34 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 10:38.51 | hirnukuono | i call my direct inbound route 152 with my cell, then transfer that call from 152 to my colleague at 134 == no problem. BUT, |
| 10:39.37 | hirnukuono | if i call a queue (10) with my cell, wait in queue, someone picks it up and then transfers it to anyone, then the person transferring is stuck with a channel open for the duration of the call. |
| 10:40.18 | hirnukuono | that person can then receive new calls from the queue or directly but our AMI-based tools show that he's busy and his call duration is the transferred call.. |
| 10:40.45 | sjs205 | Does anyone here have experience with integrating opensips and asterisk? |
| 10:41.17 | kaldemar | hirnukuono: "stuck with a channel open" <-- how did you come to this conclusion? |
| 10:41.27 | hirnukuono | aterisk -r core show channels concise |
| 10:41.41 | WIMPy | hirnukuono: Sounds like an issue in that tool, doesn't it? |
| 10:41.50 | hirnukuono | i call in, i see that there's a channel with my cell -> xxx |
| 10:41.58 | WIMPy | Is that a local channel? |
| 10:42.30 | kaldemar | hirnukuono: you'd better use a pastebin and show what really goes on. |
| 10:42.31 | hirnukuono | sec, will obfuscate the lines and send |
| 10:44.35 | hirnukuono | this is core show channels concise after my cell -> queue -> 148, and then 148 transfers to me (152), and 148 hangs up: |
| 10:44.40 | hirnukuono | Local/148@from-internal-0000015d;2!macro-dial-one!s!42!Up!Dial!SIP/152,15,trM(auto-blkvm)!MY_CELL!!!3!15!SIP/152-00000130!1365603041.1002 |
| 10:44.43 | hirnukuono | Local/148@from-internal-0000015d;1!from-internal!10!1!Ringing!AppQueue!(Outgoing Line)!152!!!3!64!SIP/fin2-0000012d!1365603041.1001 |
| 10:44.46 | hirnukuono | SIP/fin2-0000012d!ext-queues!10!12!Up!Queue!10,t,,!MY_CELL!!!3!71!Local/148@from-internal-0000015d;1!1365603034.997 |
| 10:44.49 | hirnukuono | SIP/152-00000130!macro-dial-one!s!1!Up!AppDial!(Outgoing Line)!152!!!3!15!Local/148@from-internal-0000015d;2!1365603091.1004 |
| 10:44.58 | WIMPy | ~pb |
| 10:44.58 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 10:45.06 | hirnukuono | the Ringing AppQueue line stays in for the duration of the whole call |
| 10:45.48 | hirnukuono | ... i apologize for not pb'ing it :/ lesson learned. |
| 10:45.48 | *** join/#asterisk timahvo1 (~rogue@mail.sbakenyaltd.com) |
| 10:46.00 | kaldemar | hirnukuono: no channel to SIP/148 there. that's just a local channel that stays on the path because of /n, probably. |
| 10:46.25 | kaldemar | hirnukuono: sounds like your AMI tool needs fixing. |
| 10:47.04 | hirnukuono | yesterday we got reports that the queue agents stayed busy after transferring |
| 10:47.17 | hirnukuono | that's why i'm stuck with investigating this |
| 10:47.30 | kaldemar | on "direct" transfers local channels are not used, right? |
| 10:47.37 | hirnukuono | but of course, today when we tested this, my transferer got new calls eventho my line was open.. |
| 10:48.06 | hirnukuono | i'll test it right now |
| 10:49.30 | hirnukuono | correct, no local channels when calling in directly & transferring |
| 10:53.19 | kaldemar | hirnukuono: bad parsing on the reporting tool might explain this. it may think that the existence of Local/148 means that SIP/148 is busy. the reporter might have also taken a look at the reporting tool only. who knows. |
| 10:59.03 | *** join/#asterisk Bradada (~Bradada@220-135-49-159.HINET-IP.hinet.net) |
| 10:59.24 | hirnukuono | kaldemar: thanks, i'll talk with the person doing ami stuff, the tool was originally built against 1.2 api, and we've done a lot of fixes (rolled in 1.8 past weekend) .. |
| 11:00.10 | hirnukuono | now everything works except only pstn -> queue -> transfer.. found some google links to people with similar effects but they were 2006-2009 posts.. |
| 11:03.42 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-040-087.pools.arcor-ip.net) |
| 11:05.25 | MalMen | its possible to mask asterisk to connect to a trunk as another program ? |
| 11:06.37 | WIMPy | If you're talking SIP see sip.conf. |
| 11:07.13 | kaldemar | MalMen: useragent is the parameter |
| 11:08.17 | kaldemar | MalMen: you might want to change sdpsession and sdpowner too. |
| 11:09.54 | MalMen | kaldemar useragent i already found to change on sip_general_additional.conf |
| 11:10.05 | MalMen | where are that 2 parameters ? |
| 11:10.31 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 11:10.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.80) |
| 11:10.52 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 11:11.26 | linocisco | i have asteriskNow like GUI on my QNAP builtin asterisk |
| 11:11.50 | linocisco | but I dont know how to install extra module like End Point Manager |
| 11:12.58 | kaldemar | MalMen: http://svn.digium.com/svn/asterisk/tags/11.3.0/configs/sip.conf.sample |
| 11:14.44 | ariel_ | linocisco: there is a channel for asterisknow it's #asterisknow. But in freepbx depending on your version it's via module admin, and the new End point Manager has been renamed to oss epm |
| 11:16.53 | linocisco | ariel_, yes. I checked with AsteriskNow. noone answerd yet. in freepbx , there is module admin, here is asterisk 1.4. |
| 11:17.39 | linocisco | ariel_, yes. I checked with AsteriskNow. noone answerd yet. in freepbx , there is module admin, here is asterisk 1.4. nothing i can find to install modules. I have downloaded EPM free version. tgz file. I dont know which path I should install to |
| 11:18.48 | MalMen | http://pastebin.ca/2355669 |
| 11:18.50 | MalMen | can help ? |
| 11:19.19 | MalMen | <PROTECTED> |
| 11:19.39 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:20.53 | ariel_ | linocisco: in the web under module admin there is a button that says upload module just go through there and browse for the tgz file |
| 11:21.39 | ariel_ | bbl need to drive to the office, but there is also a channel that might help out it's #freepbx |
| 11:21.53 | linocisco | ariel_, there is no module |
| 11:28.44 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-040-087.pools.arcor-ip.net) |
| 11:32.15 | *** join/#asterisk smirker (~x@101.163.95.234) |
| 11:40.41 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 12:05.32 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.85) |
| 12:07.19 | *** join/#asterisk micols (~t@shell1.rlogin.dk) |
| 12:09.49 | s-hell | Hi guys! |
| 12:10.53 | s-hell | Can anyone give me a hint. My clients should be able to send text messages to each other. Is it possible to archiv this goal with sip? If yes, how? |
| 12:13.42 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 12:19.34 | kaldemar | s-hell: yes. use sip clients that support it and use a version of asterisk that supports out-of-call messaging with sip (>= 10). |
| 12:19.38 | *** join/#asterisk Captain_Proton (~quassel@173.162.32.1) |
| 12:20.40 | s-hell | hm, ok, will take a look,. |
| 12:21.45 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:22.00 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 12:22.01 | Captain_Proton | ? for any I have a Wildcard TE122 Card w/echo hardware will it cause problem if I also turn on software echo cancelling? Will they fight making the echo worse? |
| 12:24.06 | *** join/#asterisk Cubber (~ronny@cpe-74-71-254-190.twcny.res.rr.com) |
| 12:26.43 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 12:26.54 | Greenlight | You can only select one type of echo cancel |
| 12:27.04 | Greenlight | You want to use hwec, if you have it |
| 12:28.21 | [TK]D-Fender | Captain_Proton, SWEC will never activate as long as you have HWEC. |
| 12:30.45 | Captain_Proton | [TK]D-Fender: Thanks. one more ? is there a way to reload dahdi with restarting it? I mean if I change chan-dahdi can I just rload asterisk and it will make the changes or do I need to restart dahdi |
| 12:31.03 | [TK]D-Fender | Captain_Proton, Yes you need to restart DAHDI |
| 12:31.10 | Captain_Proton | ok |
| 12:31.12 | [TK]D-Fender | actually... depends |
| 12:31.32 | [TK]D-Fender | entire signalling changes might require a channel reload (timing source, etc) |
| 12:31.33 | kaldemar | Captain_Proton: depends on what settings you change. channel definitions always require a restart. |
| 12:31.33 | Captain_Proton | I mostly changes DTMF stuff |
| 12:31.43 | [TK]D-Fender | Captain_Proton, that should be just * |
| 12:31.58 | Captain_Proton | this flex crap is killing me |
| 12:34.22 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 12:38.34 | *** join/#asterisk bjhaid (~bjhaid@197.152.30.40) |
| 12:40.38 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 12:41.24 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 12:41.24 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 12:43.47 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 12:48.43 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:55.17 | *** join/#asterisk serafie (~erin@nat/digium/x-vxhalxrlhytyfgti) |
| 12:55.49 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 12:57.17 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:05.08 | s-hell | can someone please take look at my log: http://paste.pcspinnt.de/pastebin.php?show=m43b2cbfd |
| 13:05.22 | s-hell | can't send messages via sip. i get a timeout |
| 13:05.50 | s-hell | for me, the log looks good |
| 13:12.00 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:12.01 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:13.38 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 13:21.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.80) |
| 13:32.38 | Rhomber | well I got fail2ban working with the new asterisk security stuff |
| 13:32.55 | Rhomber | ironically, i've had no break-in attempts since then to test it |
| 13:33.05 | Rhomber | but managed to ban myself quite happily lol |
| 13:37.00 | *** join/#asterisk PickledParrot (~aj@79.135.102.10) |
| 13:37.52 | Captain_Proton | here a ? I need to do a p-cap on a interface where a pri hit asterisk . How can I do that? |
| 13:39.59 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 13:55.14 | *** join/#asterisk MalMen (~MalMen@bl6-156-183.dsl.telepac.pt) |
| 13:55.49 | tzafrir | Captain_Proton, recent dahdi has support for that with some Digium cards, I believe |
| 13:55.57 | Rhomber | I also updated the wiki to include the HOWTO http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk |
| 13:57.34 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 14:01.45 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.84) |
| 14:05.02 | elguero | Captain_Proton: check into dahdi_pcap |
| 14:05.35 | *** join/#asterisk DND (5ec8071a@gateway/web/freenode/ip.94.200.7.26) |
| 14:05.37 | DND | hi guys |
| 14:06.05 | DND | my dchannel is down. im not sure what to do :( im on E1 using te121 |
| 14:06.30 | Captain_Proton | tzafrir: elguero: Thanks |
| 14:07.48 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:10.02 | DND | i tried different configurations but still not working and when i call i get: http://pastebin.com/raw.php?i=s2cC7Fka and dahdi_scan is: http://pastebin.com/raw.php?i=u2SVFpnQ |
| 14:10.17 | DND | can anyone explain hangup cause 27? |
| 14:11.39 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 14:14.12 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:14.12 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:19.06 | *** join/#asterisk PickledParrot (~aj@79.135.102.10) |
| 14:19.06 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 14:19.06 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 14:19.07 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 14:19.07 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 14:19.07 | *** join/#asterisk phunguy (~phunguy@dhcp.i-p.org.uk) |
| 14:19.07 | *** join/#asterisk thehar (~thehar@diddlebox.thehar.com) |
| 14:19.07 | *** join/#asterisk krzdrn (~sb@static-71-185-213-22.phlapa.fios.verizon.net) |
| 14:21.34 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 14:24.21 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:24.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:26.54 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 14:27.12 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 14:30.54 | MalMen | when i define externhost sip trunk fail to regist on server, what can be the problem ? :\ |
| 14:32.00 | pabelanger | Aliens |
| 14:32.12 | pabelanger | show us the problem, better then us just guessing |
| 14:32.37 | [TK]D-Fender | MalMen, Keep these in #freepbx as the solutions and places to make changes are specific since it runs your configs, not you |
| 14:32.39 | MalMen | ok, i will put here the logs |
| 14:32.48 | [TK]D-Fender | malcolmd, #freepbx <--- |
| 14:32.52 | [TK]D-Fender | MalMen, ^ |
| 14:33.05 | [TK]D-Fender | grumbles at his shoddy auto-complete |
| 14:33.25 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 14:33.38 | itgrl | is no longer away : Gone for 1 day 18 hours 9 minutes 33 seconds |
| 14:36.55 | pabelanger | itgrl: please disable your away message |
| 14:37.50 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:41.53 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 14:46.59 | *** join/#asterisk Devon_ (~Devon_@63.214.236.169) |
| 14:47.09 | itgrl | Done, I thought I already had until I came back this morning. |
| 14:53.47 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:54.01 | *** join/#asterisk rocketcitymel (~Adium@nat/digium/x-xplrvjducdxrarbk) |
| 14:57.11 | s-hell | how can i determe a module verision? |
| 15:00.02 | *** join/#asterisk bipul (1b3e9f6f@gateway/web/freenode/ip.27.62.159.111) |
| 15:15.10 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 15:15.40 | Captain_Proton | elguero: I just download a complied the latest DAHDI 2.3.6 and complied dahdi_pcap. i did not do a make install as this is a production server. So my ? is if a do a make install on DAHDI what will happen (I dont think again)? Also do I need to restart before I can run dahdi_pcap |
| 15:21.21 | elguero | Captain_Proton: well, if the version already installed did not have dahdi_pcap, you will need to update the drivers which would mean you need to schedule maintenance on the server to update the drivers to what you just downloaded and compiled |
| 15:21.27 | *** part/#asterisk bjhaid (~bjhaid@197.152.30.40) |
| 15:23.29 | Captain_Proton | elguero: Thanks that what I thought just making sure. So tonight just do "make install" and reboot so asterisk will use the new dahdi. correct? |
| 15:25.42 | igcewieling | s-hell: Asterisk modules don't really have "versions" |
| 15:26.33 | *** join/#asterisk fireman_biff (~biff@65.48.133.103) |
| 15:28.12 | elguero | Captain_Proton: That would be the easiest if you want to make sure that the drivers are unloaded and then loaded |
| 15:29.25 | fireman_biff | Hi all, I'm using asterisk 1.8 with an E1 and all calls that go out over the E1 have large periods of silence throughout the call, maybe 5 seconds of silence for every 1 second of audio. Inbound calls are fine, calls over SIP are fine, the provider says everything looks good on their end. Any ideas? |
| 15:34.35 | *** join/#asterisk RyanTG (~Thunderbi@65.100.106.194) |
| 15:40.18 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 15:44.22 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 15:48.39 | Qwell | wonders who rocketcitymel is |
| 15:51.34 | Qwell | retracts his pondering |
| 15:53.57 | Greenlight | When Asterisk starts I see the following error about failing to load a configuration file. How can I see which file it means? http://pastebin.com/uMrNyJzn |
| 15:54.48 | igcewieling | unless you are using SMDI that is a harmless message. |
| 15:55.05 | WIMPy | Not that one |
| 15:55.27 | WIMPy | Do a relaod and see if that gives better output. |
| 15:55.27 | Greenlight | Yea, the "failed to load" message ? |
| 15:55.29 | Qwell | Greenlight: run with verbose. |
| 15:55.35 | igcewieling | maybe start asterisk as asterisk -cvvv so you can actually see the stuff you need to see? |
| 15:55.45 | [TK]D-Fender | res_config_sqlite.conf.sample |
| 15:55.49 | [TK]D-Fender | res_config_sqlite3.conf.sample |
| 15:55.50 | [TK]D-Fender | ^^^ |
| 15:55.53 | Greenlight | Ah yea, sorry, being a bit dim today... two secs |
| 15:56.08 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/11/configs/ |
| 15:56.09 | vlad_starkov | Question: Anyone familiar with A2Billing? Is it possible to qualify provider trunks? |
| 15:56.27 | Greenlight | Oh.. it needs them *both*? |
| 15:56.39 | [TK]D-Fender | Registered Config Engine sqlite3 |
| 15:56.40 | [TK]D-Fender | ..................................................................................................................Failed to load configuration file. |
| 15:56.48 | [TK]D-Fender | my first guess is SQLITE3 |
| 15:57.17 | WIMPy | vlad_starkov: What does one have to do with the other? |
| 15:57.49 | vlad_starkov | WIMPy: what do you mean? |
| 15:58.05 | MalMen | anyone know a good sip provider that provide free national calls ? |
| 15:59.23 | [TK]D-Fender | MalMen, Where do you get this impression that service is free? |
| 16:00.03 | Greenlight | Verbose mode gives less dots, but no more detail... http://pastebin.com/z2hnAeAA |
| 16:00.07 | MalMen | [TK]D-Fender the service is never free, but are providers that give a good amount of minutes for a small amount of money |
| 16:00.21 | MalMen | some years ago i remember the voipbuster |
| 16:00.21 | [TK]D-Fender | MalMen, you just asked for FREE |
| 16:00.56 | [TK]D-Fender | ~itsplist-us |
| 16:00.57 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 16:01.05 | [TK]D-Fender | MalMen, Go look for yourself ^ |
| 16:01.22 | *** join/#asterisk vader- (~me@204.183.88.101) |
| 16:01.25 | vader- | hello |
| 16:01.28 | MalMen | tks :) |
| 16:01.33 | MalMen | ~itsplist-pt |
| 16:01.34 | MalMen | :P |
| 16:01.37 | MalMen | no luck |
| 16:01.42 | MalMen | ill google |
| 16:01.52 | Greenlight | My res_config_sqlite3.conf is almost same as the same one, just with cdr disabled. |
| 16:02.15 | Greenlight | I wish it would just say which configuration file it was having troubles with ;/ |
| 16:02.33 | [TK]D-Fender | Greenlight, Probably the one you DON'T have |
| 16:02.44 | [TK]D-Fender | Greenlight, so why don't you cut that short and setup BOTH |
| 16:02.48 | Greenlight | Yea, I'd agree |
| 16:03.13 | Greenlight | BUt it clearly reads res_config_sqlite3.conf okay, and the other one is the same ... |
| 16:03.23 | Greenlight | I'll copy it and see :) |
| 16:03.52 | Greenlight | Nope ... not that |
| 16:04.12 | Qwell | You need to actually have verbose logging enabled on the console. |
| 16:04.16 | Qwell | see logger.conf |
| 16:04.24 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-144-164.user.veloxzone.com.br) |
| 16:04.54 | igcewieling | thinks Greenlight is a 220V device plugged into a 120v outlet today |
| 16:05.28 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 16:05.35 | *** join/#asterisk navaismo (~navaismo@189.191.205.18) |
| 16:05.54 | Greenlight | I get verbose logging as it spams away happily after that, I just didn't pase the extra few hundred lines |
| 16:06.00 | Greenlight | *paste |
| 16:07.43 | Greenlight | Hmm.. logger.conf existing might help though |
| 16:07.52 | Greenlight | It's been one of those days... |
| 16:08.06 | MalMen | hmmm, skype have unlimited for 4.99 mounth |
| 16:08.07 | [TK]D-Fender | Greenlight, Are nickel-and-diming the shit out of your entire install? |
| 16:09.09 | Greenlight | Its a new box I've just built up, and I'm manually creating each config file so it's as trimmed down as it can be and I know exactly what's running and loaded |
| 16:09.30 | drmessano | We don't talk about Skype in here |
| 16:09.33 | drmessano | Skype is evil |
| 16:09.38 | drmessano | Much like telnet |
| 16:09.39 | [TK]D-Fender | Greenlight, So do we. So for Asterisk in not loaded or running :) |
| 16:10.28 | Greenlight | I must confess that normally I take the lazy approach and do make samples, then edit what I need |
| 16:11.06 | *** part/#asterisk rocketcitymel (~Adium@nat/digium/x-xplrvjducdxrarbk) |
| 16:11.12 | Greenlight | Or clear sip.conf and extensions.conf completely and start them freshly |
| 16:11.44 | Greenlight | Now that logger.conf exists it's much happier and loads up and registers it's peers like it should |
| 16:15.52 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 16:16.43 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 16:20.30 | slav3_kitten | so if i'm not mistaken here. each call needs 2 RTP ports right? |
| 16:20.55 | Greenlight | If the call consists of an incomming and outgoing channel, yes |
| 16:21.04 | [TK]D-Fender | slav3_kitten, Or more. |
| 16:21.24 | Greenlight | A "call" to say, an application would only need 1, unless I'm mistaken |
| 16:22.18 | slav3_kitten | [TK]D-Fender, if i have 6 phones, 2 inbound did's and each of the 6 phones can make outbound calls at the same time. what's the maximum number of RTP ports i need |
| 16:22.57 | slav3_kitten | 100 should cover it right? |
| 16:23.00 | [TK]D-Fender | slav3_kitten, What is your real goal here? |
| 16:23.22 | slav3_kitten | limiting the number of holes i need to poke in my wall of fire |
| 16:23.46 | Greenlight | I would consider the whole range as a single hole |
| 16:24.04 | slav3_kitten | aka trying to fix the phone rtp issue i think is actually an isp issue but they are convinced it's on my end |
| 16:24.19 | slav3_kitten | since sip is still broken |
| 16:24.25 | *** join/#asterisk lvlinux (~n1gg@c-50-142-165-230.hsd1.tn.comcast.net) |
| 16:24.27 | slav3_kitten | iax2 works fine however |
| 16:24.58 | Greenlight | iax2 is much better at transversing NAT. SOme "sip aware" routers screw with things too. |
| 16:25.04 | slav3_kitten | and i've heard that the cisco routers have issues with udp port ranges |
| 16:25.35 | Greenlight | We've a range of 10k UDP ports open on our ASA's |
| 16:25.36 | slav3_kitten | so i was going to make a wad of static port routes to my asterisk server |
| 16:25.51 | Greenlight | Ewww |
| 16:26.01 | slav3_kitten | and 10k statements stuck into the router 1 by 1 is a lil unrealistic |
| 16:26.15 | slav3_kitten | i have all the cisco alg shit disabled |
| 16:26.28 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.124.245) |
| 16:26.39 | Greenlight | It's a Cisco ASA you're using? |
| 16:27.08 | slav3_kitten | cisco 2600 series to a 3600 series switch |
| 16:27.15 | slav3_kitten | or is it 3500 |
| 16:27.24 | Greenlight | The catalyst ? |
| 16:27.27 | slav3_kitten | yea 3500 series switch sorry |
| 16:27.41 | slav3_kitten | nods at Greenlight |
| 16:27.56 | Greenlight | But that's a switch not a router ? |
| 16:28.09 | slav3_kitten | one second i need to go shoot a rat |
| 16:28.11 | Greenlight | Ahh... is this ACL ? |
| 16:28.18 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 16:28.34 | slav3_kitten | yea acl on the 2621 |
| 16:28.39 | lvlinux | the 3600 series switches do routing too i think. |
| 16:28.49 | igcewieling | slav3_kitten: The most common SIP/NAT issues are caused by people making it WAY TOO COMPLICATED. |
| 16:29.32 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 16:29.56 | Greenlight | Why not disable the ACL's for testing, and check it works? Is the issue not more likely to be on the router, if anywhere ? |
| 16:30.40 | slav3_kitten | ok rat is dead |
| 16:30.55 | lvlinux | slav3_kitten: what did u shoot it with? |
| 16:31.02 | slav3_kitten | .22lr subsonic |
| 16:31.14 | lvlinux | hehe i love those things |
| 16:31.17 | slav3_kitten | out of a 1950's bolt action rifle |
| 16:31.27 | lvlinux | the subsonic i mean, not the rats... lol |
| 16:31.44 | lvlinux | i have one of those. Stevens brand. |
| 16:31.56 | slav3_kitten | i like rats, as pets. not as big fucking garden eating yard terrorists |
| 16:32.34 | lvlinux | "yard terrorists" that's funny. But in all fairness you don't know what a "yard terrorist" is until you've had groundhogs :-) |
| 16:32.49 | slav3_kitten | lvlinux, we have those too |
| 16:33.08 | slav3_kitten | we have rats, snakes, groundhogs, moles, and crayfish |
| 16:33.19 | Greenlight | Get a cat. Rat problem sorted. |
| 16:33.19 | igcewieling | What is this "yard" thing you speak of? Is that the green ground which it is someone else's problem to take care of? 8-) |
| 16:33.35 | lvlinux | oh hehe. Yeah they made a crater in our yard about 10 feet wide and 6 deep. |
| 16:33.40 | slav3_kitten | it's an acre and a half out in the country |
| 16:33.45 | lvlinux | u must live in the country like us. |
| 16:34.19 | lvlinux | Greenlight: cats can't keep up when you have a lot of the buggers... |
| 16:35.24 | slav3_kitten | anyhow Greenlight i'm 90% sure it's not my hardware as i am unaware of anything changing. had shit working just fine until the static IP i pay 10 a month for suddenly changed now i've got no-way calling, sip signaling works just fine but once someone answers i've got no voice either direction. and sometimes the asterisk box keeps ringing but the tone of ring changes |
| 16:36.14 | slav3_kitten | AH HA HA HA the mail truck just got stuck in the washed out driveway |
| 16:36.18 | slav3_kitten | falls over laughing |
| 16:36.21 | lvlinux | all your other network access seems fine? Just not the RTP? |
| 16:36.21 | Greenlight | slav3_kitten: How very odd. Did you question your ISP on their defenition of "static ip" ? :) You've not got extenip set to the old IP or something do you / |
| 16:36.22 | lvlinux | lol |
| 16:38.42 | slav3_kitten | pretty much lvlinux, and Greenlight first thing i did was change every instance of externip |
| 16:39.29 | Greenlight | What does a SIP trace show ? |
| 16:39.50 | lvlinux | stupid question but have you rebooted your router (to reset the NAT tables or whatever)? |
| 16:40.02 | slav3_kitten | yep i did that |
| 16:40.04 | Greenlight | Yea, pfsense can be bad for that |
| 16:40.10 | slav3_kitten | Greenlight, um, wish i could remember |
| 16:40.31 | slav3_kitten | also \o/ my car part arrived |
| 16:40.34 | Greenlight | At the very least pipe it to a file and go a grep for the old IP address.... |
| 16:40.47 | Greenlight | Just to be sure :) |
| 16:40.55 | slav3_kitten | oh i checked for that Greenlight |
| 16:41.04 | slav3_kitten | got the new IP in there |
| 16:41.57 | Greenlight | It's not something really silly like a typo on the new externip address. I've had like a "2" and "5" wrong way around before and at a glance the brain doesn't notice it... hours of my life lost on that |
| 16:42.09 | slav3_kitten | like i said signaling works fine, iax2 works fine, sip is broken. what i believe may have happened is that the SIP alg on the ubiquiti "modem" i have (a nanostation 2) got turned back on |
| 16:42.32 | lvlinux | sounds like it's time for some tcpdump. |
| 16:42.41 | slav3_kitten | which is what was the source of my problems originally |
| 16:42.59 | lvlinux | signaling works fine? I thought SIP was the signaling? |
| 16:43.01 | Greenlight | WHoever invented SIP ALG should get the same treatment as that rat... |
| 16:43.11 | lvlinux | yah buddy |
| 16:43.28 | slav3_kitten | lvlinux, i should say port 5060 seems to be unfucked up |
| 16:43.55 | slav3_kitten | < < in no way a sip or asterisk wizard |
| 16:44.05 | slav3_kitten | < < < often gets terms wrong |
| 16:44.09 | Greenlight | Can you get into that modem and ensure it's turned off |
| 16:44.26 | Greenlight | The SIP ALG that is |
| 16:44.28 | slav3_kitten | Greenlight, funnily enough. no |
| 16:44.41 | lvlinux | ?? |
| 16:44.44 | slav3_kitten | that's all remote admined by a single guy at the ISP that knows how to do it |
| 16:44.52 | slav3_kitten | whom is on vacation again |
| 16:44.56 | Qwell | get a new modem? |
| 16:44.57 | lvlinux | oooooooooooohhhhhhhhhh..... |
| 16:44.59 | Qwell | get a new ISP? |
| 16:45.04 | Greenlight | Yea, new ISP |
| 16:45.08 | lvlinux | get a sledge hammer |
| 16:45.09 | slav3_kitten | only game in town |
| 16:45.10 | Greenlight | That gets my vote |
| 16:45.21 | lvlinux | is it DSL or what? |
| 16:45.23 | slav3_kitten | it's this WISP, 56k, or sat |
| 16:45.41 | slav3_kitten | since sat a 56k suck a box of dicks... i have the wisp option |
| 16:45.41 | Greenlight | At the very least, blag the admin password out of them ^^ |
| 16:46.10 | Qwell | move >.> |
| 16:46.13 | Greenlight | lol |
| 16:46.14 | slav3_kitten | also i kinda want to burn down comcast whom assured us we could get cable modem service before we put an offer in on the house |
| 16:46.22 | slav3_kitten | Qwell, not an option |
| 16:46.26 | Qwell | sure it is |
| 16:46.26 | igcewieling | slav3_kitten: sometimes VoIP is NOT the correct solution. |
| 16:46.27 | slav3_kitten | like at all |
| 16:46.31 | Qwell | I'll trade you houses. |
| 16:46.32 | igcewieling | accept it and move on |
| 16:46.49 | Greenlight | Presumably the WAF comes into play |
| 16:46.56 | Qwell | also, why not just VPN your SIP traffic? |
| 16:46.58 | Greenlight | Wife-acceptance-factor that is |
| 16:47.08 | lvlinux | make them setup ur modem as a bridge if possible. |
| 16:47.13 | Qwell | or like TLS, or one of the many options to tell the ALG to screw off |
| 16:47.27 | slav3_kitten | igcewieling, what do you propose the correct solution is to calling my lover in the united kingdom without costing myself an arm and a leg |
| 16:47.28 | Greenlight | Yea, run it in bridge mode if you can, although this sounds more like a wireless router than a modem |
| 16:47.55 | lvlinux | skype? lol no i didn't just say that |
| 16:48.01 | slav3_kitten | Greenlight, it is a wireless router which is supposedly in bridge mode. but the tech whom can change thing is on vacation |
| 16:48.17 | igcewieling | slav3_kitten: there may not be a solution. VoIP requres something close to an OK network connection. You do not have that. |
| 16:48.18 | slav3_kitten | lvlinux, you're going to hell |
| 16:48.33 | igcewieling | Like trying to make bread with no flour. No matter how hard you try, it won't work. |
| 16:48.44 | slav3_kitten | igcewieling, before the IP change i had an OK connection |
| 16:48.55 | Greenlight | You've two options as I see it: 1. VPN. 2. Get a sip trace from *both* sides of the connection and compare. See if that "modem" is fucking with it. |
| 16:48.57 | slav3_kitten | it actually as i previously said was working perfectly |
| 16:49.05 | igcewieling | But what you HAVE is taken large amounts of people's time on this channel to help you with something which won't work. |
| 16:49.41 | slav3_kitten | actually i thought i was conversing in a channel which is mostly inactive. you don't need to participate if you feel it's a waste of time igcewieling |
| 16:49.56 | Greenlight | Why not use IAX if that works ? |
| 16:50.24 | slav3_kitten | also since no one actually answered my previous question of "what's a decent amount of rtp channels for something i want to test" ... |
| 16:50.52 | slav3_kitten | Greenlight, because i have 100 dollars in flowroute which is my sip itsp due to their cheap rates to my girlfriend's cell |
| 16:51.15 | igcewieling | slav3_kitten: how many simultaneous calls? |
| 16:51.18 | lvlinux | decent amount can vary -- if you are just doing one channel you can just open a few (2? 4?), but that is going to make your setup more complex any may not help |
| 16:51.24 | slav3_kitten | up to 6 igcewieling |
| 16:51.35 | igcewieling | slav3_kitten: the answer is 12 ports |
| 16:51.42 | Greenlight | slav3_kitten: Get a VPS for $5/month. IAX2 to there, SIP out. Win. |
| 16:51.44 | igcewieling | 24 if you have video |
| 16:52.02 | lvlinux | 12? really? I thought u just needed one for TX and one for RX? |
| 16:52.15 | Greenlight | You said 6 extensions didn't you ? |
| 16:52.16 | lvlinux | Greenlight: where can you get a VPS for $5 month? |
| 16:52.18 | slav3_kitten | lvlinux, it will make it more complex and may not help. but it's something to test out |
| 16:52.21 | igcewieling | lvlinux: he wants 6 calls at 2 ports per call, you do the math |
| 16:52.23 | slav3_kitten | i did Greenlight |
| 16:52.33 | lvlinux | oh 6 calls yes 12 ports. |
| 16:52.37 | Greenlight | http://www.ovh.co.uk/vps/ |
| 16:53.05 | slav3_kitten | igcewieling, see i just asked very simply is it 2 ports per channel, and got "yea but could be more" then no information to quantify "more" |
| 16:53.40 | igcewieling | slav3_kitten: the "more" is usually "video", but if you transfer it could be more or less. Just open up 100 ports and don't worry about it. |
| 16:53.48 | slav3_kitten | also thanks i had not considered a VPS, never even occurred to me |
| 16:54.08 | slav3_kitten | thanks igcewieling, sorry i "wasted" your time and the time of so many others here |
| 16:54.09 | Greenlight | You can also VPN to the VPS |
| 16:54.28 | Greenlight | Anyways... time to leave the office and get some dinner. Laters! |
| 16:54.35 | igcewieling | slav3_kitten: you have wasted much more of YOUR time. |
| 16:54.38 | slav3_kitten | nom well Greenlight |
| 16:54.50 | slav3_kitten | my time is pretty much worthless so it's not like that's really a concern |
| 16:55.03 | slav3_kitten | i also count this as my fucked up ISP wasting my time |
| 16:57.39 | lvlinux | hey slav3 what exactly are you doing? are you connecting to an ITSP with multiple trunks or ? |
| 16:58.51 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 17:01.50 | slav3_kitten | lvlinux, i've got (to save money) flowroute as my outbound ITSP and voip.ms as my inbound with 2 DIDs |
| 17:02.20 | lvlinux | k |
| 17:05.56 | lvlinux | hey slav_kitten is it possible that you arebeing NAT'ed on the ISP side? |
| 17:07.09 | SeRi | slav3_kitten: I do the same. |
| 17:07.22 | SeRi | slav3_kitten: flowroute is cheaper in the us that voip.ms |
| 17:07.27 | slav3_kitten | yep |
| 17:07.31 | SeRi | s/that/than/ |
| 17:08.11 | slav3_kitten | lol |
| 17:08.15 | slav3_kitten | i knew what you meant |
| 17:09.54 | SeRi | guys is it possible to have one context use two different caller ids base on the dialing rule IE: 1NXXNXXXXXX will display one caller id and 91NXXNXXXXXX will display another |
| 17:10.35 | [TK]D-Fender | SeRi, Your dialplan does whatever you tell it to |
| 17:14.50 | MalMen | [TK]D-Fender when i set dynamic host only the sapo sip still getting timeouts |
| 17:15.00 | MalMen | voipbuster work allright anyway |
| 17:17.19 | navaismo | 1,110 euros for a LPIC-Junior certification its fine? To me its quite expensive but not sure |
| 17:19.40 | *** join/#asterisk joesmoe (~joesmoe@c-76-112-110-95.hsd1.mi.comcast.net) |
| 17:20.06 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 17:22.21 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-izdsbpysxlrlhozf) |
| 17:25.08 | slav3_kitten | SeRi, absolutely |
| 17:25.33 | *** join/#asterisk _Corey_ (~corey@64.215.11.114) |
| 17:26.08 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.144) |
| 17:26.35 | SeRi | no matter hwat I do is setting my caller id to one and not the other |
| 17:26.43 | SeRi | even if I seperate the context |
| 17:27.34 | navaismo | Check if your provider allow the overwrite CID |
| 17:27.43 | SeRi | navaismo: flowroute does |
| 17:29.59 | SeRi | here is my pb http://pastebin.com/30sdaBGc |
| 17:30.11 | SeRi | no matter what I do it uses the first cid and not the second |
| 17:30.29 | SeRi | I cant understand how if they are two different context |
| 17:33.14 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 17:33.47 | [TK]D-Fender | SeRi, that PB proves nothing. We don't see what your CALL is using. |
| 17:35.28 | navaismo | well Seri you have a Goto to the 1${EXTEN} |
| 17:35.58 | navaismo | so I GUESS(GUESS) that is sending to the top of the pb and setting the first CID |
| 17:36.10 | navaismo | but as fender said show us the call |
| 17:40.41 | SeRi | guys |
| 17:40.45 | SeRi | http://pastebin.com/hy09MPgC |
| 17:41.02 | SeRi | I left the # there so you can see what exactly is happening |
| 17:41.10 | SeRi | I hope I dont get screwed :/ |
| 17:41.37 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:41.37 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:41.45 | navaismo | check your Goto |
| 17:42.01 | SeRi | navaismo: my goto should stay in the same context no? |
| 17:42.13 | SeRi | not jump to another |
| 17:42.17 | SeRi | or amI confused? |
| 17:42.27 | SeRi | s/amI/am I/ |
| 17:42.35 | navaismo | doesn't matter, your issue is the destination of that GOTO |
| 17:43.47 | navaismo | basically your goto takes the _NXXNXXXXXX numbers and convert it in _1NXXNXXXXXX |
| 17:44.41 | SeRi | navaismo: yes because my itsp only takes 1NXXNXXXXXX |
| 17:44.54 | SeRi | if 1 is not present it rejects the outbound call. |
| 17:46.16 | SeRi | I think I am confused. so what should it look like? should it not continue to the top and just continues to dial? |
| 17:47.46 | [TK]D-Fender | SeRi, You have failed to learn the basic of INCLUDE's as well and decided it's goo to jump around with a merged scope. |
| 17:48.01 | navaismo | Well i dont know your logic or waht you are trying to do, I only pointing why its using the same CID |
| 17:48.15 | SeRi | [TK]D-Fender: lol got it. |
| 17:48.26 | SeRi | navaismo: I think I figured it out. Thanks All. |
| 17:48.38 | [TK]D-Fender | SeRi, You showed us 2 contexts... and were using a THIRD |
| 17:48.46 | [TK]D-Fender | (or more) |
| 17:50.00 | *** join/#asterisk nightrid3r (~michel@62.205.66.100) |
| 17:58.24 | lvlinux | i have a question (pretty elementary i think): how do i strip a digit from an extension? |
| 17:59.32 | [TK]D-Fender | lvlinux, You don't. You can strip it from the VALUE of a variable and then do something with that remaining value. |
| 17:59.35 | lvlinux | in other words if i have an extension _4NXXXXXX how can i send that out to be dialed as just NXXXXXXX |
| 17:59.50 | lvlinux | well that's what i meant |
| 17:59.52 | itgrl | lvlinux> I do it with the dial command |
| 17:59.53 | itgrl | exten => s,n,Dial(${TRUNKY}/${MACRO_EXTEN:1},,wW) |
| 17:59.55 | [TK]D-Fender | lvlinux, $(VAR:numcharstostrip:offset} |
| 18:00.04 | itgrl | the :1 strips off the left most digit. |
| 18:00.30 | [TK]D-Fender | lvlinux, ${VAR[:numcharstostrip[:offset]]} |
| 18:00.34 | *** part/#asterisk fireman_biff (~biff@65.48.133.103) |
| 18:00.37 | lvlinux | ah ok thanks |
| 18:01.54 | SeRi | Thank you all. I dont work with asterisk day in and out so I tend to get very rusty.... as all ways [TK]D-Fender bluntness gets me to start remembering shit... navaismo thanks for the help as well. |
| 18:02.30 | lvlinux | so i can do this: exten => _2NXXXXXX,1,Dial(SIP/mytrunk/${EXTEN}:1,,) |
| 18:02.38 | lvlinux | is that correct? |
| 18:02.58 | lvlinux | and it will strip the 2? |
| 18:03.02 | navaismo | ${EXTEN:1} |
| 18:03.09 | lvlinux | oh |
| 18:03.12 | SeRi | ^^ |
| 18:03.19 | SeRi | as navaismo pointed :) |
| 18:04.42 | lvlinux | alrighty thanks! |
| 18:05.53 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 18:14.07 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:18.53 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:18.53 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:41.28 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 19:04.36 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:08.07 | FLeiXiuS | How does asterisk keep track of which RTP stream belongs to which SIP registration? |
| 19:08.12 | FLeiXiuS | ERm, or where* |
| 19:08.45 | WIMPy | RTP streams do not belong to registrations. |
| 19:10.36 | FLeiXiuS | Erm, I meant sip channel. I'm having a problem where a user is using a mobile device and when switching from wifi to 3g, the connection is reacquired on the mobile side, but Asterisk isnt aware of the tear down. |
| 19:10.56 | FLeiXiuS | So when the device retransmits RTP, asterisk believes its for the first call, rather the 2nd reinvite. |
| 19:11.21 | FLeiXiuS | Then I get a lack of RTP timeout etc |
| 19:12.24 | WIMPy | It should accept a new stream after a reinvite. |
| 19:12.31 | WIMPy | What version? |
| 19:12.37 | FLeiXiuS | 1.8 |
| 19:13.17 | WIMPy | I'd sugest to try a current version. |
| 19:13.27 | FLeiXiuS | I thought so too, I figured it may have been the defaulted pedantic setting being too restrictive on the reinvite. |
| 19:15.33 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 19:18.00 | FLeiXiuS | WIMPy, My mistake, its not reinviting, it's creating a SIP session |
| 19:18.24 | WIMPy | What does that mean? |
| 19:18.27 | FLeiXiuS | Which then throws the new RTP stream from that session - into the old call. |
| 19:19.24 | WIMPy | Sounds like the opposite of what you asked about before. |
| 19:20.43 | FLeiXiuS | After understanding it a bit more, heres the workflow. I create a call from wifi, move to 3g. A new call is created on 3g, but the RTP is going to the old wifi call. Then the new 3g channel is killed by asterisk because of a lack of RTP |
| 19:21.49 | FLeiXiuS | I know its going to the old call, because I record all of the calls made and the streams being appended to the first recording. |
| 19:22.31 | WIMPy | Are you sure you see a new call? Sounds wrong. |
| 19:23.02 | WIMPy | But still, |
| 19:23.05 | FLeiXiuS | I am, both are being recorded, I see the new registration/invite. I see the new call torn down afterwards as well due to a lack of RTP |
| 19:23.10 | WIMPy | ~upgrade asterisk |
| 19:23.10 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 19:23.14 | WIMPy | That will also give you ICE support. |
| 19:23.15 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.smb.curriegrad2004.ca) |
| 19:24.23 | FLeiXiuS | I'll upgrade on my dev box in order to rule that out. |
| 19:26.45 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.smb.curriegrad2004.ca) |
| 19:34.45 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 19:42.59 | *** join/#asterisk afink (4172ce22@gateway/web/freenode/ip.65.114.206.34) |
| 20:02.46 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.smb.curriegrad2004.ca) |
| 20:03.16 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.smb.curriegrad2004.ca) |
| 20:08.21 | vader- | man i really wish i could hunt down the call quality issue this office is having with their system |
| 20:08.35 | vader- | Their setup looks like this: |
| 20:09.02 | *** join/#asterisk francisvgarcia (~francisvg@186.33.104.127) |
| 20:09.25 | vader- | Pholycom 670 Phones <---> Asterisk 11 <---> Xorcom Astribank <----> Comcast POTS lines via ATA/VOIP Gateways <---> Coaxial |
| 20:09.35 | vader- | Inter office calls are clear |
| 20:09.52 | vader- | Comcast has "tested" their pots lines and say everything is clear |
| 20:10.06 | vader- | The office will make calls and most of the time they are crystal clear |
| 20:10.24 | vader- | but they have been getting an abnormal amount of noise, hollow sound, etc. |
| 20:10.47 | vader- | i have tried hooking a regular phone up to the POTS lines and calling |
| 20:10.51 | vader- | i don't really hear the same thing |
| 20:11.01 | vader- | any recommendations on what i can check? |
| 20:11.49 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 20:14.50 | [TK]D-Fender | lthe ATA side is unclear |
| 20:14.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.83) |
| 20:15.04 | [TK]D-Fender | Astribank already hits TDM. Where does an ATA fit in after that? |
| 20:15.32 | WIMPy | It's connected to an ATA? |
| 20:15.38 | vader- | ya |
| 20:15.46 | WIMPy | The normal standardstupidity these days. |
| 20:16.32 | WIMPy | At least you get the option here to use BRI instead. |
| 20:17.07 | vader- | https://www.dropbox.com/s/rwm6ufaatvm2fbd/2013-03-05%2001.47.46-2.jpg |
| 20:19.00 | vader- | they had locked into this contract with comcast before they told me they wanted a new telephone system |
| 20:19.08 | vader- | so they have 8 POTS lines with them |
| 20:19.17 | drmessano | Your antivirus is out of date |
| 20:19.42 | vader- | haha their "server closet" is in shambles |
| 20:19.48 | WIMPy | s still waiting for the image to appear. |
| 20:19.49 | vader- | you don't want to see what the rest of the room looks like |
| 20:19.53 | drmessano | Sorry, couldnt resist |
| 20:20.00 | vader- | they use it for storage of everything |
| 20:20.25 | drmessano | If there is still a box for Sav 10 laying around, yeah |
| 20:20.59 | vader- | it was funny they called comcast to come in |
| 20:21.04 | vader- | to check the lines |
| 20:21.18 | vader- | the guy somehow blew the firmware out of the xorcom unit |
| 20:21.31 | vader- | freaked out and started calling places to try and repair it. instead of me |
| 20:21.57 | vader- | so im not sure who he called |
| 20:22.14 | vader- | but some how he got the phone number to the "owner of xorcom" he tells me |
| 20:22.41 | vader- | and the owner of xorcom says the astribank is not made to be used with asterisk and that i don't have it setup right |
| 20:23.18 | vader- | it's hilarious |
| 20:23.24 | drmessano | HAHAHHA |
| 20:23.24 | vader- | this tech was a fool |
| 20:23.32 | vader- | ya he tells the business manager this |
| 20:23.36 | vader- | she is reeming me a new one |
| 20:23.39 | vader- | becuase i don't know what im doing |
| 20:23.55 | vader- | i don't know how he blew the firmware out of the unit which had been working perfectly for 9 months |
| 20:24.10 | vader- | anyways you goto xorcom's website: http://www.xorcom.com/products/product-catalog/telephony-interfaces/astribank-usb-channel-banks |
| 20:24.11 | drmessano | Because with a name like "ASTRIbank", it sounds like something designed for a Cisco PBX |
| 20:24.15 | vader- | and click on architecture |
| 20:24.28 | vader- | Asterisk server |
| 20:24.28 | vader- | hehe |
| 20:25.41 | drmessano | So they use Comcast for their "IT" ? |
| 20:25.51 | vader- | this women gets frantic |
| 20:25.55 | vader- | and just calls everyone |
| 20:25.59 | vader- | anyone who will help her |
| 20:26.08 | drmessano | Comcast DOES do small business IT work |
| 20:26.11 | drmessano | Just not well |
| 20:26.20 | vader- | she has experienced that twice this month |
| 20:26.26 | vader- | she called comcast in to repair her internet |
| 20:26.32 | vader- | i told her to unplug her router power |
| 20:26.35 | vader- | and plug it back in |
| 20:26.39 | vader- | she called comcast instead |
| 20:26.49 | vader- | who rewired her router, network, etc |
| 20:26.53 | vader- | luckily i take pictures |
| 20:27.01 | vader- | i put all the cables back in place where they were suppose to be |
| 20:27.04 | vader- | and rebooted the router |
| 20:27.07 | drmessano | I had a company with two locations. The firm I was with supported the HQ, and the satellite office 3 hours away was supported by Comcast, right after they first started doing it |
| 20:27.08 | vader- | tada worked |
| 20:27.20 | vader- | she doesn't listen |
| 20:27.29 | drmessano | They were just terrible |
| 20:27.41 | vader- | but she is driving me nuts with this hollow sound |
| 20:27.45 | vader- | noise/static/etc |
| 20:27.54 | vader- | everyday she calls me about it |
| 20:27.57 | vader- | and i can't hunt it down |
| 20:28.38 | drmessano | Hell, I had a Comcast tech out a coworkers house last night. "What kind of Router is that?" "A Linksys, WRT54G" "Yeah, they're not bad". Old, but 10x better than the shit you're peddling Comcrap |
| 20:28.50 | drmessano | I was offended, personally, by his idiocy |
| 20:28.52 | drmessano | So I feel you |
| 20:29.37 | *** join/#asterisk serafie (~erin@24.214.158.242) |
| 20:30.20 | drmessano | He basically told me "They're not bad" because he had nothing else to contribute to the conversation. He could have easily said "Yeah, they're Blue" and it would have been just as intelligent |
| 20:30.48 | drmessano | </rant> |
| 20:31.36 | leifmadsen | :D |
| 20:40.09 | *** join/#asterisk themrrobert (themrrober@unaffiliated/themrrobert) |
| 20:41.39 | themrrobert | so what might possibly locking? It seems my PBX is randomly locking up for a second or two, pausing all calls briefly, and starting again with a small burst of traffic. I am unable to test this with DEBUG_THREADS, because by the time the system even gets close to partially loaded, the call quality is so terrible across every second that i can't really check whats locking |
| 20:44.00 | mjordan | themrrobert: sometimes you can tell from just a DEBUG log. Also: are you using CDRs? And do you have batch mode enabled? |
| 20:44.56 | *** join/#asterisk maslowbeer (~maslowbee@static-96-251-72-193.lsanca.fios.verizon.net) |
| 20:46.25 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 20:47.01 | themrrobert | i tried enabling debug in the logger file but it didn't actually write debug info to the log, am i missing something? and batch mode is not enabled, looks like i should try that, i didn't know i could do that. on a different system i set up csv logging with a cron job to update the db, but we do use CDR's. i haven't noticed any high mysql cpu usage, but i'm sure its used a lot. |
| 20:47.05 | themrrobert | how often does batch update the cdr? |
| 20:47.27 | smkelly | file: hi |
| 20:47.47 | newtonr | themrrobert: re the debug not being written to log file make sure debug isn't set to 0 in asterisk.conf |
| 20:47.51 | mjordan | that's configurable. Take a look at the cdr.conf.sample file for examples. You can make it batch on every single record if you want. What batch mode does however is take the processing of the CDR off of the threads servicing the channel |
| 20:48.57 | smkelly | Is it possible to configure res_digium_phones in a db backend instead of having to put stuff in that .conf file fore each phone? |
| 20:49.08 | themrrobert | right, @mjordan, can i set that in cdr_mysql.conf? not using the regular cdr.conf |
| 20:49.30 | mjordan | no, cdr_mysql configures the MySQL CDR backend driver. cdr.conf configures the actual CDR processing engine. |
| 20:49.56 | mjordan | and this is just speculation. Without actually analyzing what the bottlenecks are, it's guesswork. This is just one thing to try. |
| 20:52.51 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:52.51 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:54.46 | *** join/#asterisk arnotixe (~Thunderbi@190.130.243.195) |
| 20:55.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:56.12 | arnotixe | hi all I've got asterisk setup on alpine linux (no timezone tree, only a /etc/TZ file saying "COT5"). "date" in the alpine linux shell shows correct local time, but the asterisk cli shows GMT, and CDR also shows GMT. Any suggestions on how to fix this? |
| 20:56.17 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:56.17 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:56.21 | *** join/#asterisk Cartes (cartes@116.193.170.3) |
| 20:56.46 | Cartes | hi all |
| 21:01.09 | newtonr | smkelly: Not for res_digium_phone.conf directly. https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration does provide you with some info on what to do if your SIP peers are stored using realtime (ARA). See the table under Line Configuration Options. |
| 21:01.46 | malcolmd | static realtime, that is, not dynamic. ^ |
| 21:02.08 | *** join/#asterisk nephfl (6c09e362@gateway/web/freenode/ip.108.9.227.98) |
| 21:02.11 | nephfl | hello, if doing an http cid lookup, what is the format of the return value? |
| 21:04.18 | smkelly | thanks |
| 21:04.43 | malcolmd | smkelly: np |
| 21:05.11 | malcolmd | nephfl: i don't think anyone can answer that question given the information you've provided |
| 21:05.17 | [TK]D-Fender | nephfl: From who? This isn't part of * |
| 21:06.03 | nephfl | I have a data source and a web app, I can return whatever I would like for the given get request, but I don't know what asterisk is expecting...is it just the name, or the name/number or something else |
| 21:06.23 | [TK]D-Fender | nephfl: * isn't expecting anything... * doesn't talk to HTTP for CID |
| 21:06.55 | nephfl | I'm not using a provider, I am trying to configure my own source...and google has docs for configuring a request, but not for formatting a return value |
| 21:07.55 | nephfl | you mean the cid source lookup module is not part of core asterisk but part of freepbx? |
| 21:08.08 | [TK]D-Fender | Clearly |
| 21:08.20 | [TK]D-Fender | the word MODULE alone should have given it away |
| 21:08.27 | [TK]D-Fender | SIP has callerid |
| 21:08.39 | [TK]D-Fender | Analog FSK on DAHDI has callerid. |
| 21:08.44 | nephfl | ah, sorry about that, wrong channel sortof then...unless you happen to know about third party lookup sources return values |
| 21:08.49 | [TK]D-Fender | Digital signalling in Q.931 has callerid |
| 21:09.08 | [TK]D-Fender | nephfl: #freepbx <- |
| 21:09.09 | nephfl | sip is a "module" as well, but I wouldn't assume it wasn't part of asterisk's core functions |
| 21:11.12 | [TK]D-Fender | nephfl: This is not. Those in #freepbx may have a better idea. |
| 21:11.32 | nephfl | ok, np, thanks |
| 21:11.41 | [TK]D-Fender | nephfl: Also sample ones should give you enough to test direct and see for yourself |
| 21:12.21 | nephfl | that is what I was thinking, I was going to have a look at the dialplan and see what it is expecting |
| 21:13.32 | arnotixe | how can I start asterisk with another timezone set? Tried TZ="Europe/Oslo" /usr/sbin/asterisk but seems it still uses GMT... |
| 21:14.23 | *** join/#asterisk micols (~t@shell1.rlogin.dk) |
| 21:16.49 | [TK]D-Fender | arnotixe: TZ is a function of your server itself, and the only other place is voicemail.conf |
| 21:16.58 | [TK]D-Fender | arnotixe: As it translates time back |
| 21:19.47 | arnotixe | ok so asterisk timezone is setup in voicemail.conf? |
| 21:22.14 | arnotixe | hmm maybe I should just hack some php code that rewrites the CDR stamp to correct local time... |
| 21:23.25 | [TK]D-Fender | arnotixe: You should process it in whatever actually looks at it, but it should remain agnostic |
| 21:23.48 | [TK]D-Fender | arnotixe: voicemail is one place that you perceive TZ's |
| 21:27.21 | themrrobert | thanks mjodan i appreciate it i will let you know how that goes. |
| 21:30.21 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 21:31.49 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 21:31.49 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 21:33.30 | *** join/#asterisk jsjc (~Adium@249.Red-83-46-236.dynamicIP.rima-tde.net) |
| 21:33.54 | arnotixe | [TK]D-Fender: yep I can understand the benefits of asterisk always giving out UTC time, that way it _always_ has to be localized, and it's universal all over the globe. I have only two sites on 2 different continents, and it's actually kind of good that they show the same time in raw logs :D |
| 21:35.00 | [TK]D-Fender | arnotixe: And who's looking there? :) |
| 21:35.37 | arnotixe | is having a quick look at the server Master.csv |
| 21:41.30 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 21:52.43 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 22:12.05 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 22:14.36 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.71) |
| 22:21.01 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 22:24.01 | *** join/#asterisk joesmoe (~joesmoe@c-76-112-110-95.hsd1.mi.comcast.net) |
| 22:32.05 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.183) |
| 22:32.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.183) |
| 22:44.36 | *** join/#asterisk deviantlinux (~ryan_gpx@unaffiliated/deviantlinux) |
| 22:48.34 | deviantlinux | Question on Voicemails -- if I want to move voicemails off an older system to a newer, I can just copy all files from /var/spool/asterisk/voicemail right? |
| 22:49.21 | navaismo | i choose yes |
| 22:54.29 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 22:54.29 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 22:58.56 | *** join/#asterisk fkurkowski (~fkurkowsk@177.18.101.137) |
| 23:05.35 | *** join/#asterisk Jagster` (~chatzilla@unaffiliated/jagster/x-9084543) |
| 23:11.19 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 23:15.08 | *** join/#asterisk netman (netman@249.140.20.95.dynamic.jazztel.es) |
| 23:24.33 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:25.10 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 23:28.11 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 23:49.42 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 23:50.18 | *** part/#asterisk deviantlinux (~ryan_gpx@unaffiliated/deviantlinux) |
