| 00:04.05 | pabelanger | markit: http://svn.asterisk.org/svn/asterisk/trunk/doc/lang/language-criteria.txt |
| 00:05.02 | markit | pabelanger: thanks a lot! |
| 00:05.04 | adnc | WIMPy, do you know of a documentation or example? |
| 00:05.40 | pabelanger | markit: np |
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| 00:13.22 | x86 | pabelanger: ah... no, i did not... thanks! |
| 00:13.49 | x86 | pabelanger: you know how I can set the ringer volume from the provisioning files? |
| 00:14.01 | x86 | pabelanger: I want to max out the ringer volume and the handset volume |
| 00:14.13 | x86 | it always resets to mid-volume every reboot |
| 00:14.53 | pabelanger | x86: Off the top of my head no |
| 00:14.59 | x86 | hmm |
| 00:15.04 | pabelanger | I'm sure there is a settings for it |
| 00:15.16 | x86 | yeah, I just can't read the admin guide from the console |
| 00:15.22 | x86 | it's only available as a PDF :( |
| 00:16.24 | pabelanger | Ya, I'm not sure why they don't provide .txt or .html |
| 00:16.30 | x86 | ah! |
| 00:16.47 | x86 | <volume voice.volume.persist.handset="1"> |
| 00:16.50 | x86 | w00t! |
| 00:17.45 | x86 | that takes care of that, but not the ringer |
| 00:19.18 | x86 | se.pat.ringer.2.inst.12.value="-11" ? |
| 00:19.24 | x86 | will changing that change the volume? |
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| 00:29.20 | x86 | how well does 1.8.0 support SRTP with a Polycom phone (if at all) |
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| 00:33.01 | pabelanger | x86: It should work, 1.8.0 is the first release with SRTP |
| 00:33.28 | x86 | cool |
| 00:34.48 | x86 | will have to try it |
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| 00:35.41 | x86 | hmm, so I can provision the ring tone, and make it non-user-changeable, but I can't figure out how to provision the ringer volume... |
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| 01:25.25 | dev_astguru | hi |
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| 01:26.52 | dev_astguru | anyone having issues with voicemail server running on vmware? |
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| 01:42.18 | jblack | drmessano: heh |
| 01:47.43 | dev_ast | hi JBlack |
| 01:48.02 | jblack | Hi dev_ast |
| 01:48.26 | dev_ast | I have voicemail server running under VMware Esx server |
| 01:49.05 | dev_ast | whenever there are 5-7 simultenous calls, CPU usage goes 100% |
| 01:49.17 | jblack | Sorry. I lost interest the second you said vmware. |
| 01:49.26 | dev_ast | 30% asterisk 50% sendmail |
| 01:49.32 | dev_ast | oh ok.... |
| 01:50.58 | dev_ast | So, what version of asterisk are you using? |
| 01:51.06 | jblack | 1.4.something |
| 01:51.13 | dev_ast | I am still using 1.4.24 considering it as most stable |
| 01:51.45 | dev_ast | did u try to implement Shared Line Appearance |
| 01:51.49 | jblack | Nope. |
| 01:52.31 | dev_ast | it seems that Asterisk 1.6 has SLA support on it, but couldn't make it working |
| 01:54.55 | jblack | Might be. I had a variety of mild problems with 1.6, so I backed down to 1.4 |
| 01:55.18 | jblack | I think 1.8 is just around the corner, if not out yet |
| 01:55.28 | dev_ast | could you please share some of the issues on 1.6 |
| 01:55.56 | jblack | Yeah, looks like 1.8 came out on 10/26. |
| 01:56.04 | dev_ast | in fact i was planning to upgrade all of my production servers from 1.4 to 1.6 |
| 01:56.28 | dev_ast | yes 1.8 came out on 10/21 |
| 01:56.37 | jblack | Nothing useful to really say about it, it's been a long time since I did 1.6, and whatever i saw was probably long fixed |
| 01:57.55 | dev_ast | I am afraid to go with 1.8 for production. |
| 01:58.18 | dev_ast | Hopefully 1.6 doesn't have critical issues on SIP |
| 01:59.03 | WIMPy | dev_ast: As long as I don't use srtp it's working. Or at least has been doing so for three days. |
| 01:59.39 | drmessano | I noticed a few issues with 1.8 as well.. like maybe there's a memory leak somewhere |
| 02:00.09 | jblack | That wouldn't be a surprise. |
| 02:00.43 | dev_ast | WIMPy: Thanks for the information. I will look on that |
| 02:01.06 | WIMPy | The srtp thing is pretty funny. I get audio in both directions, but after about 5 seconds asterisk crashes. |
| 02:01.48 | WIMPy | Interstingly the phone displays an insecure connection. |
| 02:01.57 | dev_ast | huh!! weird |
| 02:01.58 | drmessano | Without the lecture on Linux memory usage, I noticed my home box with a 1GB RAM is using much more RAM than before, which may be by design, but I am also consuming swap and never did with any 1.6 release or 1.4 release.. Also noticed I go unresponsive after a few days, and I either need to restart (if I notice it) or eventually there's a crash then a restart from safe_asterisk that clears it |
| 02:02.28 | WIMPy | Yes, it's using more by design. |
| 02:03.22 | WIMPy | I'll keep an eye on the memory size. |
| 02:03.23 | dev_ast | drmessano: it could be due to lots of modules embedded in asterisk 1.8 |
| 02:03.36 | drmessano | Lots of modules embedded? |
| 02:04.05 | dev_ast | when you compile asterisk, customize it for your requirement |
| 02:04.14 | dev_ast | make menuselect |
| 02:04.17 | drmessano | I don't embed any of the modules |
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| 02:04.31 | drmessano | I compile all the same as I ever have |
| 02:04.52 | drmessano | But that shouldn't crash asterisk |
| 02:05.51 | dev_ast | i will try to install Asterisk 1.8 on dell poweredge r710 |
| 02:06.00 | dev_ast | and see if it does funny like you said |
| 02:07.56 | ChannelZ | Mine is hillarious |
| 02:08.24 | dev_ast | what happened ChannelZ? |
| 02:08.42 | ChannelZ | Cracking jokes left and right |
| 02:09.02 | drmessano | Oh, you installed app_funny? |
| 02:09.16 | dev_ast | nice one drmessano |
| 02:09.25 | dev_ast | :) |
| 02:09.47 | ChannelZ | It's a lot of fart and poop humor, but still.. |
| 02:11.04 | ManxPower | I think I will wait for 1.8.0.20 8-) |
| 02:11.43 | dev_ast | Asterisk is not good for the use of Voicemail and IVR. Please comment on it |
| 02:12.10 | dev_ast | most importantly Playback() function |
| 02:12.46 | drmessano | What??? |
| 02:13.03 | drmessano | Where do you get your information? |
| 02:13.29 | dev_ast | i m getting lots of complaint from my customers |
| 02:13.40 | dev_ast | they are getting jitters while retrieving voicemails |
| 02:14.22 | dev_ast | when i say jitters, i mean delay in playback |
| 02:14.22 | drmessano | Sounds like your servers are either (1) poorly configured virtual machines (2) underpowered or (3) your bandwidth sucks |
| 02:14.43 | dev_ast | that's not true |
| 02:15.32 | drmessano | [22:11] <dev_ast> Asterisk is not good for the use of Voicemail and IVR. Please comment on it <-- That's not true either, and what you described is not a "known issue" with Asterisk or else it would have been addressed |
| 02:15.38 | dev_ast | as virtual center shows resources is perfectly assigned |
| 02:16.16 | dev_ast | i have VLAN assigned for the voice with QOS enabled network |
| 02:16.37 | drmessano | That's meaningless.. Your management console isn't going to tell you that you're running a REALTIME application in a non-so-realtime environment |
| 02:16.53 | drmessano | Which would explain just what you're describing |
| 02:17.50 | dev_ast | so what's your point. I shouldn't be using virtual machine for asterisk |
| 02:19.38 | ChannelZ | Or maybe that you shouldn't be complaining about it |
| 02:19.51 | drmessano | It sounds like you're using paravirtualization and not a fully virtualized environment |
| 02:21.03 | dev_ast | I am using fully virtualized machine... I am using SLES to host asterisk as voicemail server |
| 02:22.35 | dev_ast | Is there anyone using asterisk on Virtual Machine. You might be right that VMs are not smart enough to handle realtime app |
| 02:22.58 | ChannelZ | well voip requires fairly decent timing, if you're getting stutters it implies not a very good clock source or possibly packets disappearing into nowhere |
| 02:23.32 | dev_ast | i used wireshark to trace the issue |
| 02:23.59 | WIMPy | voip is not a realtime thing. It only is if you have hardware interfaces. - BUT - You DO need a reliable timing source. And VMs seem to have issues there. |
| 02:24.43 | pabelanger | ManxPower: There won't be a 1.8.0.20, just 1.8.20 |
| 02:24.49 | ChannelZ | It doesn't matter if you have hardware interfaces or not - if your machine can't count and spits out packets too slowly you're going to have problems |
| 02:24.50 | dev_ast | that's what i thought WIMPy. But couldn't find the real solution to solve that |
| 02:25.34 | dev_ast | may be i can talk with VMware guys, but not sure they can do anything with this |
| 02:26.58 | dev_ast | network bandwidth is not my issue. I have plenty of network bandwidht |
| 02:27.17 | dev_ast | so it should be VM |
| 02:28.33 | ManxPower | pabelanger, not unless they changed the release procedures again |
| 02:29.04 | pabelanger | yes, it changed |
| 02:29.06 | ManxPower | we had 1.6.0.x 1.6.2.x 1.6.3.x as Asterisk versions |
| 02:29.21 | pabelanger | 1.8.x, 1.10.x, etc |
| 02:29.40 | ManxPower | can you cite a reference for changing it back to the old versioning? |
| 02:30.16 | pabelanger | ManxPower: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 02:31.02 | ManxPower | pabelanger, thank you. |
| 02:31.10 | ManxPower | they need to stop changing the versioning layout |
| 02:33.27 | pabelanger | _should_ be the last time |
| 02:35.24 | dev_ast | pabelanger, thanks for the information |
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| 04:49.47 | ChannelZ | Crap, the time change is tonight isn't it.. |
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| 06:21.16 | mmlj4 | it is |
| 06:23.36 | baileyx | hey there, my asterisk install on (on ubuntu) works well, i am trying to install asterisk-gui but am having a strange problem. it successfully starts a http service on port 8088, but when i try to browse to http://theip/asterisk/static/config/ it says '404 not found' â when I try http://theip/asterisk/static/ '403 permission denied' is there a log for this asterisk/http service so i can troubleshoot |
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| 06:25.00 | p3nguin | ~asterisk-gui |
| 06:25.00 | infobot | [~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0. For support go to #asterisk-gui |
| 06:25.33 | baileyx | thanks |
| 06:25.58 | p3nguin | I can't guarantee success in that channel, either, but it has to be better than this one. |
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| 06:57.31 | ChannelZ | are you going to http://theip-asterisk-static:8080/config ? |
| 06:58.06 | ChannelZ | (or whatever, but the important part being :8080 after the ip/hostname) |
| 07:00.03 | baileyx | yes, the port is there. now it's working for me . i had to make a symlink as it appears the server was simply looking in the wrong folder |
| 07:03.26 | baileyx | now it's tuck in a Checking write permission for gui folder, but this seems like a resolvable permissions issue |
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| 07:16.20 | dandate2 | is caller id spoofing illegal yet? |
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| 07:28.47 | x86 | no |
| 07:28.50 | x86 | nub |
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| 10:05.55 | seanjohn | any way to get asterisk to show a timestamp in the console? |
| 10:08.20 | seanjohn | I see this feature request from 2003: https://issues.asterisk.org/view.php?id=107 did it ever happen? |
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| 13:34.05 | robl^laptop | howdy all! |
| 13:35.02 | robl^laptop | anyone know if the Vestec Speech Recognition Engine supports Asterisk 1.8 yet or if support is planned? |
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| 13:58.18 | pabelanger | robl^laptop: Best to contact Digium directly |
| 13:59.53 | robl^laptop | pabelanger: yeah, I'll prolly contact them on Monday. I was just checking to see if anyone here was familiar with it. |
| 14:00.43 | pabelanger | I don't see anything related to 1.8.0 on http://downloads.digium.com/pub/telephony/vestec/ |
| 14:00.48 | pabelanger | So I would say no ATM |
| 14:02.15 | atan | If you have automon => *1 can you set it to save files into a set folder? like, /dev/null if there is no accountcode, or /recordings/{accountcode}/ ? |
| 14:02.19 | robl^laptop | pabelanger: that's what I noticed, as well, hence my asking for confirmation. thanks. |
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| 14:21.35 | atan | Hey guys I am having the same issue this fella is, any ideas? http://forums.digium.com/viewtopic.php?f=1&t=75808&p=149702&hilit=greeting&sid=be4468a8ccb3eec952e1f2625d3a72cf#p149702 |
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| 14:56.01 | atan | I missed the flag part. Figues =) We're good now |
| 15:01.45 | atan | I seem to be missing something with this thinger, exten => _NXXXXXX,n,Set(CDR(accountcode=${SIPPEER(accountcode)})) |
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| 15:02.33 | [TK]D-Fender | atan: brackets in the wrong place. |
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| 15:03.00 | [TK]D-Fender | atan: completefunctionreference=value |
| 15:03.57 | atan | [TK]D-Fender, the brackets don't look out of place to me though? |
| 15:04.46 | [TK]D-Fender | atan: = comes after you have completed the function. in yours it does not |
| 15:05.21 | atan | exten => _NXXXXXX,n,Set(CDR(accountcode=${SIPPEER(accountcode)})) should be exten => _NXXXXXX,n,Set(CDR(accountcode${SIPPEER=(accountcode)})) ? |
| 15:05.25 | atan | is so lost right now |
| 15:05.35 | [TK]D-Fender | atan: atan CDR is not closed before the "=" <--------- |
| 15:06.23 | atan | Ahh, so like exten => _NXXXXXX,n,Set(CDR(accountcode)=${SIPPEER(accountcode)}) |
| 15:06.34 | [TK]D-Fender | atan: THERE you go |
| 15:06.52 | atan | Hmm. I must have missed something else. Not setting the accountcode still =\ |
| 15:07.20 | [TK]D-Fender | atan: it is calling SET and passing the value you expect from SIPPEER? |
| 15:07.24 | atan | Set("SIP/7940-023009f0", "CDR(accountcode)=") in new stack |
| 15:07.24 | atan | <PROTECTED> |
| 15:08.43 | atan | I think I got it, one sec ;D |
| 15:08.47 | atan | I feel so foolish sometimes |
| 15:08.58 | [TK]D-Fender | atan: First the CDR shoudl already have the accountcode from your peer... it shouldn't have to be assigned manually... |
| 15:09.20 | [TK]D-Fender | atan: forgot to set it in the peer I'm betting.. or typo'd |
| 15:09.28 | atan | _NXXXXXX,1,Set(SIPID=${SIPCHANINFO(peername)}) |
| 15:09.32 | atan | _NXXXXXX,n,Set(CDR(accountcode)=${SIPPEER(${SIPID}:accountcode)}) |
| 15:09.44 | atan | Is what I'm going for but no doubt I goofed up there somewhere |
| 15:10.19 | [TK]D-Fender | atan: Again, this isn't something you're supposed to have to set in the first place. |
| 15:10.26 | [TK]D-Fender | atan: pastebin the call and your peer |
| 15:13.38 | atan | [TK]D-Fender, http://pastie.org/private/olh0yg6fubvd7nttr4wu7q |
| 15:16.33 | atan | [TK]D-Fender, we're good here now. You were 100% right, it was a stupid typo on my part. |
| 15:18.18 | atan | [TK]D-Fender, can we back up a few steps here. I think you hinted I was using more lines of config than I needed when I had exten => _NXXXXXX,1,Set(SIPID=${SIPCHANINFO(peername)}) exten => _NXXXXXX,n,Set(CDR(accountcode)=${SIPPEER(${SIPID}:accountcode)}) |
| 15:18.24 | [TK]D-Fender | atan: Also you should not have to set it at all. |
| 15:18.47 | atan | Let me remove those lines & see what happens |
| 15:19.00 | [TK]D-Fender | atan: accountcode=1234567989 <---- this should be already passed on as the starting value for CDR(accountcode) |
| 15:20.12 | atan | Well I'll be. Sweet deal. |
| 15:21.16 | atan | [TK]D-Fender, any recommendations on 911 service? |
| 15:21.25 | atan | Should I flag that call somehow do it's never dropped? |
| 15:23.03 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 15:25.11 | [TK]D-Fender | atan: You normally only set the CDR direct if that value is variable... |
| 15:25.37 | atan | [TK]D-Fender, so in the case of flagging inbound calls then? |
| 15:25.53 | [TK]D-Fender | atan: For instance |
| 15:27.39 | atan | [TK]D-Fender, is there ever a time where billsec would be > than duration ? |
| 15:27.52 | atan | And is there a billing interval in some config file? |
| 15:28.58 | [TK]D-Fender | nope |
| 15:29.00 | phix | hmmmm |
| 15:29.08 | phix | hey [TK]D-Fender, you are one of my heroes |
| 15:29.23 | [TK]D-Fender | atan: how YOU process the facts is up to you after the fact... it should only store the reality |
| 15:30.00 | phix | <3 |
| 15:30.28 | atan | [TK]D-Fender, inside cdr.conf if I change unanswered = yes to see unanswered calls (so I can capture the ring times) it doesn't insert the accountcode |
| 15:30.55 | phix | I am having issues with fax dectection |
| 15:31.02 | phix | it is picking up faxes as talk not fax |
| 15:31.07 | phix | even when I wait 4 secs |
| 15:31.33 | phix | should I just compile NVDetectSomething or use the inbuilt stuff which seems to be fialing? |
| 15:32.48 | [TK]D-Fender | atan: Sounds like it isn't inheriting until bridged... doesn't sound too logical to me... might be considered a "bug" |
| 15:40.15 | [TK]D-Fender | time to head out, back later |
| 15:46.28 | *** join/#asterisk genxweb (~genxweb@c-174-59-109-188.hsd1.pa.comcast.net) |
| 15:48.51 | genxweb | I have a question in my extensions_custom.conf file I created a new context and it works fine. My question is I name my trunks the tr- then extension number. I want to have it grab the extension of the user placing the call and choose that trunk for the outbound call. This is what I ahve tried exten => _X.,3,Dial(SIP/{AMPUSER}/${EXTEN},120,tr) but it does nto work I even tried wit ha $ in |
| 15:48.51 | genxweb | front of teh {AMPUSER} |
| 15:49.32 | genxweb | for right now I am trying to get it to say SIP/extension#/outbound_number |
| 15:53.30 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 16:03.27 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 16:03.39 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 16:09.58 | *** join/#asterisk thansen (~thansen@112.sub-97-162-92.myvzw.com) |
| 16:11.26 | atan | what is DBput in 1.8? |
| 16:16.47 | atan | pbx_extension_helper: No application 'DBput' for extension =S |
| 16:20.03 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 16:20.24 | ketas | hm |
| 16:20.36 | ketas | hugs asterisk |
| 16:24.31 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 16:27.43 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 16:43.16 | *** join/#asterisk adolfomaltez (~taro@190.86.146.150) |
| 16:43.20 | pabelanger | atan: Set(DB(blah)=1234) |
| 16:43.49 | atan | Thank you. =) |
| 16:46.04 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 16:46.10 | z4nD4R | hi all |
| 16:46.56 | ketas | hi z4nD4R |
| 16:48.25 | z4nD4R | om my asterisk server are loged 2 users ( 1000 and 1001 ) with TLS... when 1000 ( on the same machine as server ) call 1001 all is good, but if 1001 call 1000 the call is not established.. any ideas? |
| 16:50.58 | pabelanger | z4nD4R: How about showing us the problem? |
| 16:51.23 | z4nD4R | pabelanger: on softphone show failed.. |
| 16:51.31 | adolfomaltez | hi z4nD4r, the client 1000 and the server overlaps SIP port, i think |
| 16:51.54 | pabelanger | ~collectdebug |
| 16:51.54 | infobot | somebody said collectdebug was a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 16:51.58 | pabelanger | z4nD4R: ^^ |
| 16:52.52 | adolfomaltez | when i use ekiga softphone in the same Asterisk server, i have to change te SIP port for ekiga |
| 16:53.10 | z4nD4R | adolfomaltez: you right |
| 16:53.11 | z4nD4R | 1001/1001 158.193.85.60 D 5061 Unmonitored |
| 16:53.17 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 16:53.22 | z4nD4R | port is 5061 |
| 16:55.39 | z4nD4R | adolfomaltez: i use sip-communcator... |
| 16:57.02 | z4nD4R | adolfomaltez: you use template to each softphone? |
| 16:58.31 | *** join/#asterisk tootai (~quassel@keewi.tootai.net) |
| 17:00.49 | adolfomaltez | z4nD4r: i use ekiga. Ekiga listen for inbound sip calls in udp port 5060. Asterisk uses the same port udp/5060. Then you have to change the port for ekiga (in your case for sip-comunicator) for inbound SIP calls. you can use the udp/5070 for your sip phone in the SAME Asterisk server |
| 17:01.20 | *** join/#asterisk sbszulu (~dundubala@41.16.40.204) |
| 17:03.28 | z4nD4R | adolfomaltez: in my case i use port 5061 but i undertandt what you whant to say.... so i should be cange port on sip-communikator... right? |
| 17:05.03 | adolfomaltez | yes, change the port in the sip-cominicator, and try again |
| 17:07.07 | z4nD4R | adolfomaltez: but... proxy or regostrat... i thing no one |
| 17:07.24 | adolfomaltez | none |
| 17:08.09 | adolfomaltez | in ekiga is: "listen port" |
| 17:08.27 | z4nD4R | adolfomaltez: this options is not on sip-communikator |
| 17:09.23 | phix | nn |
| 17:09.24 | phix | ,3 |
| 17:09.25 | phix | <3 |
| 17:10.12 | genxweb | I am trying to get ths to work exten => _X.,3,Dial(SIP/tr-${AMPUSER}/${EXTEN},120,tr) it returns |
| 17:10.12 | genxweb | SIP/tr-/717xxxxxxx|120|tr but not the ampuser variable. I have this on my extensions_custom.conf file |
| 17:10.59 | adolfomaltez | z4nD4r: run as root # netstat -u4l --numeric |
| 17:11.07 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:11.12 | adolfomaltez | only for verify in what port is asterisk listen |
| 17:12.07 | z4nD4R | adolfomaltez: http://pastebin.com/W0pffVym |
| 17:13.39 | adolfomaltez | ok, the 5060 port is used by Asterisk OR sip-comunicator , the port is overlap |
| 17:14.03 | adolfomaltez | the 5061 port is not in use, for inbound sip connections |
| 17:14.41 | adolfomaltez | you can change the SIP port in asterisk instead of sip-cominicator, |
| 17:14.52 | z4nD4R | adolfomaltez: i connect trought TLS - 5061 => 5060 is used by asterisk |
| 17:16.23 | adolfomaltez | sorry, i not used TLS before, but the 5061 port is closed |
| 17:17.10 | z4nD4R | adolfomaltez: netstat -ln | grep 5061 |
| 17:17.10 | z4nD4R | tcp 0 0 0.0.0.0:5061 0.0.0.0:* LISTEN |
| 17:17.44 | adolfomaltez | i'm apologize |
| 17:18.23 | z4nD4R | ;) |
| 17:19.36 | z4nD4R | you idea is, i thing right because user on asterisk si connect on port 5061... that this port is overlaped...no??? |
| 17:19.50 | z4nD4R | <PROTECTED> |
| 17:19.50 | z4nD4R | Name/username Host Dyn Forcerport ACL Port Status |
| 17:19.50 | z4nD4R | 1000/1000 158.193.85.34 D 55607 Unmonitored |
| 17:19.50 | z4nD4R | 1001/1001 158.193.85.60 D 5061 Unmonitored |
| 17:19.50 | z4nD4R | 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] |
| 17:20.04 | z4nD4R | my ip is .60 |
| 17:23.30 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 17:24.18 | adolfomaltez | sorry z4nD4r, at this point i have no idea :( |
| 17:24.46 | z4nD4R | heh..oki... thx |
| 17:25.57 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 17:32.24 | *** part/#asterisk dlirit (~lirant@80.74.100.10) |
| 17:33.10 | pabelanger | ~collectdebug |
| 17:33.10 | infobot | collectdebug is probably a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 17:33.18 | pabelanger | z4nD4R: ^ pastebin your debug log |
| 17:34.28 | *** join/#asterisk A4eT3 (~chatzilla@cpe-76-183-56-158.tx.res.rr.com) |
| 17:37.55 | z4nD4R | pabelanger: i muss on cca 1hour go away.... but when i go back ... a paste this log |
| 17:42.23 | p3nguin | manxpower: You asked at least once if the IAX2 registration/authorization bug (the one that matches the last peer entry) was ever fixed... did you get an answer? |
| 17:46.29 | dev-ast | SIP might be the better option over IAX2. SIP is the industry standard and it's easier to get the support for SIP than IAX2 |
| 17:47.09 | WIMPy | You are aware, that it's usually the worst option that becomes standard? |
| 17:47.39 | florz | as in: asterisk? =:-) |
| 17:47.53 | WIMPy | ... |
| 17:48.09 | WIMPy | florz: What is the better option? |
| 17:48.24 | ManxPower | p3nguin, no, but I don't really care. I have not used IAX2 in years. |
| 17:48.27 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 17:48.37 | ManxPower | the workaround is simple |
| 17:48.48 | florz | well, yeah, I suppose it's the best option also, as bad as it is ;-) |
| 17:49.05 | ManxPower | um, the worst option is SIP, that is why everyone uses it. |
| 17:49.19 | WIMPy | florz: That's what I think. |
| 17:50.19 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 17:50.52 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 17:51.17 | ketas | is sip bad? |
| 17:51.28 | WIMPy | yes |
| 17:51.30 | ManxPower | p3nguin, I know the bug existed from Asterisk 0.65 all the way up to 1.4.23 |
| 17:51.47 | ManxPower | ketas, it is a poorly defined protocol, nat unfriendly |
| 17:52.07 | florz | ManxPower: what is poorly defined about it? |
| 17:52.16 | ketas | everything is nat unfriendly |
| 17:52.29 | ManxPower | florz, for one thing there are at least a dozen RFCs that "define" SIP as it is used today. |
| 17:52.53 | ManxPower | ketas, NO, not everything embeds the IP address as part of the data of the packet, making it impossible for normal NAT to work |
| 17:52.58 | florz | ManxPower: that sounds more like extensibility than poorly defined, I'd think |
| 17:53.12 | ketas | hm, well |
| 17:53.42 | florz | and as far as NAT is concerned: well, practically that's a problem, but in principle that's really NAT's fault |
| 17:54.23 | ketas | with ipv6, nat will go away? |
| 17:54.35 | WIMPy | In principle, yes, but NAT has been a common situation for about 15 years. |
| 17:54.36 | ManxPower | To be fair RTP/SDP is what is nat unfriendly. |
| 17:54.55 | ManxPower | WIMPy, good thing they didn't design SIP to work with the standards of the day, huh? |
| 17:55.27 | WIMPy | No, RTP is ok, it's the way it's set up via SIP. |
| 17:55.51 | ManxPower | H323, SCCP/Skinny, MGCP, and SIP all suffer from similar issues WRT NAT |
| 17:56.13 | ketas | wait, can't do sip behind nat? |
| 17:56.31 | WIMPy | ketas: In practice, you can. |
| 17:56.45 | genxweb | ketas I do all the time |
| 17:56.49 | ManxPower | ketas, Only if your endpoints have special SIP NAT support. You cannot do it without special NAT support for SIP in the endpoints or in the NAT router. |
| 17:56.54 | genxweb | you jsu have to open the ports |
| 17:57.01 | genxweb | and do a port fowarding or 1 to 1 nat |
| 17:57.02 | WIMPy | But you need to disable reinvites, which would have been a clever idea. |
| 17:57.13 | ManxPower | genxweb, your endpoints have special NAT support. |
| 17:57.17 | genxweb | no |
| 17:57.19 | genxweb | they dont |
| 17:57.26 | genxweb | you cna do it with a simple asterisk trick |
| 17:57.26 | ManxPower | genxweb, are you using Asterisk? |
| 17:57.31 | genxweb | yep asterisk |
| 17:57.40 | ManxPower | genxweb, Dude, Asterisk is an endpoint and Asterisk has special NAT support. |
| 17:57.46 | ketas | nat ruins everything |
| 17:58.00 | genxweb | http://www.digitaloffensive.com/2010/05/overcoming-sip-over-nat/ |
| 17:58.06 | genxweb | there that explaisn hwo to do it |
| 17:58.35 | genxweb | bbl got to go help the kids |
| 17:58.39 | ketas | i love to see that one day, typical home router routes ipv6 and is fw only |
| 17:58.44 | genxweb | if some one knows the answer to this issue please let me know |
| 17:58.46 | ManxPower | genxweb, see how that talks about special settings for NAT. |
| 17:58.51 | ManxPower | You have just proved my point |
| 17:58.58 | genxweb | yeah but that is a simpel config in inside of asterisk |
| 17:59.07 | genxweb | open ports |
| 17:59.07 | genxweb | done |
| 17:59.17 | ManxPower | I didn't say it was not simple. I said the endpoint has to have special SIP NAT support built in. |
| 17:59.44 | genxweb | I am trying to get ths to work exten => _X.,3,Dial(SIP/tr-${AMPUSER}/${EXTEN},120,tr) it returns SIP/tr-/717xxxxxxx|120|tr but not the ampuser variable. I have this on my extensions_custom.conf file |
| 17:59.47 | ManxPower | They all do these days, but it is NOT part of any SIP RFC as far as I know |
| 18:00.01 | ketas | recently found http://www.bastard.net/~kos/pf-voip.html |
| 18:00.02 | genxweb | see if anyof you can help em with that |
| 18:00.02 | ManxPower | genxweb, try asking on a channel that supports FreePBX/Trixbox |
| 18:00.09 | genxweb | I have |
| 18:00.12 | ketas | haven't tested |
| 18:00.23 | genxweb | lets forget the fact it is freepbx and put it in the extensions.conf gile |
| 18:00.29 | genxweb | same sisue |
| 18:00.34 | genxweb | same issue* |
| 18:00.40 | ManxPower | genxweb, AMPUSER is NOT an Asterisk variable |
| 18:01.04 | ManxPower | AMPUSER is a variable created by FreePBX |
| 18:01.36 | tyman | what are the opinions here of switchvox? Does anyone know how switch box can track states of calls delegated to adhearsion? |
| 18:01.39 | ManxPower | no, dialedparties.agi is also not part of the standard Asterisk, it is something from FreePBX |
| 18:03.04 | ukine_work | i lost the link for setting up gvoice with freepbx. anybody got it? |
| 18:04.06 | ManxPower | HAHAHAHA! |
| 18:04.17 | ukine_work | nvm, found it on the wiki |
| 18:04.21 | ManxPower | Freepbx question here on #asterisk. Very funny. |
| 18:05.02 | ukine_work | thanks for your help, ManxPower |
| 18:05.19 | ManxPower | ukine, maybe asking on the FreePBX channel would have been more productive? |
| 18:06.42 | ukine_work | um the settings i want to change are part of asterisk and not necessarily specific to freepbx? |
| 18:07.37 | ManxPower | ukine, virtually anything you want to do is specific to FreePBX, including the actual configuration files you edit. |
| 18:08.11 | ukine_work | thank you, THAT was helpful |
| 18:08.22 | ManxPower | For example, FreePBx will overwrite extensions.conf on every reload. |
| 18:08.43 | ManxPower | or it used to, I don't know about today -- that is freepbx specific. |
| 18:10.31 | ukine_work | still does |
| 18:10.44 | ManxPower | Asterisk does not do that -- i.e. that is a FreePBX specific thing. |
| 18:10.52 | ukine_work | yup |
| 18:11.32 | ketas | crying after lost your complex dial plan |
| 18:11.51 | ketas | s/lost/you lost/ |
| 18:12.09 | ketas | eh? |
| 18:12.11 | ketas | :) |
| 18:12.22 | ketas | helpful |
| 18:12.36 | ketas | s/.// |
| 18:12.42 | ketas | :P |
| 18:12.49 | ManxPower | My point stands. virtually anything you want to do is specific to FreePBX, including the actual configuration files you edit. |
| 18:17.52 | p3nguin | To accept registration from these random devices, I just need a peer definition without a secret? |
| 18:18.11 | ManxPower | p3nguin, ?? |
| 18:19.04 | p3nguin | manxpower: I think it was you that mentioned something the other day about allowing the random devices to register and to make calls on your system, routing any call to a prerecorded message. |
| 18:19.22 | ManxPower | p3nguin, no, allowing random passwords |
| 18:19.39 | ManxPower | the regname still has to match AFIK |
| 18:19.52 | p3nguin | oh, hmm |
| 18:19.56 | ManxPower | but I bet all the toll frauders try extension 10 or 100 first |
| 18:20.13 | p3nguin | Registration from "2403943889" |
| 18:20.24 | p3nguin | Registration from "2478091497" |
| 18:20.48 | ManxPower | create a peer with that, do not have a secret line. (don't have secret= as that requires NO password, not "any password") |
| 18:20.58 | p3nguin | They usually hit me with 6-10 digit numbers when trying to register. |
| 18:21.14 | ManxPower | p3nguin, interesting. the ones I see all start out with 2-digit extensions |
| 18:22.09 | p3nguin | I used to get attacks which would try 3-digits, often start at 100 and run up to 200 or 300 before failing and leaving. |
| 18:23.09 | p3nguin | Haven't seen any of those for a while, but I still get random attempts often trying to register 10-digit peers. |
| 18:24.00 | p3nguin | I always think they look like phone numbers, so I've actually called a few of them to see if I could figure anything out... never had any luck. |
| 18:25.47 | p3nguin | That last attempt was from 82-194-76-83.housing.hostalia.com. |
| 18:26.23 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 18:27.17 | *** join/#asterisk roe (~roe___@unaffiliated/roe) |
| 18:28.33 | genxweb | <PROTECTED> |
| 18:28.57 | p3nguin | Why did you reinvent the wheel? fail2ban does that just fine. |
| 18:29.17 | genxweb | does it do it for SIP ? |
| 18:29.24 | p3nguin | yes |
| 18:29.25 | genxweb | I know it worked on web , ssh |
| 18:29.30 | genxweb | never had it work for me on SIP |
| 18:29.33 | p3nguin | It reads asterisk's log. |
| 18:29.56 | p3nguin | ~fail2ban |
| 18:29.56 | infobot | i guess fail2ban is a program to ban people using iptables based on information in logs: http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk |
| 18:31.04 | genxweb | interesting |
| 18:31.26 | p3nguin | If you use pf (not on Linux), I'm sure it can be adapted easily enough. |
| 18:34.07 | atan | For some reason when I connect calls one person can send audio, but the other person cannot send audio |
| 18:34.17 | atan | I'm thinking it's a codec issue? Just wondering where I should start? |
| 18:34.25 | atan | It's a linksys ATA adapter on the end with the issue |
| 18:34.41 | p3nguin | Sounds like a typical SIP NAT problem. |
| 18:35.02 | atan | NAT Mapping Enable: change to "Yes": ? |
| 18:36.18 | *** join/#asterisk nicoAMG (~nicoAMG@201.237.49.131) |
| 18:36.22 | atan | No cigar on that one |
| 18:37.00 | p3nguin | Typically, you set NAT traversal to NO on the device and use Asterisk's built-in NAT settings for the peer. |
| 18:37.40 | atan | Within sip.conf I have nat=yes |
| 18:37.53 | ketas | hmm |
| 18:37.59 | p3nguin | In [general] or in the peer for the device? |
| 18:37.59 | ketas | btw, is the linksys ata good? |
| 18:38.13 | atan | p3nguin, peer for the device |
| 18:38.24 | p3nguin | Is Asterisk behind NAT? |
| 18:38.26 | atan | ketas, give me a few hours to tell you |
| 18:38.37 | ketas | :P |
| 18:38.48 | atan | p3nguin, it could be. It's sitting at the datacentre... which has some funky network scheme... |
| 18:39.02 | atan | p3nguin, I have another device (Cisco 7940) connected up which makes/gets calls just fine |
| 18:39.05 | p3nguin | Does it have a public IP address on the interface? |
| 18:39.10 | atan | p3nguin, it does |
| 18:39.23 | p3nguin | It probably doesn't have NAT, then. |
| 18:39.47 | ketas | but is firewalled? |
| 18:40.04 | p3nguin | Is the 7940 behind NAT? Is it on the same LAN as that ATA? |
| 18:40.19 | atan | The 7940 is on the same network as the ATA |
| 18:40.29 | p3nguin | Did you forward any ports at the firewall of that LAN where the phone and ATA are? |
| 18:41.01 | atan | No ports have been forwarded on the home network here, no |
| 18:41.39 | p3nguin | Does the router/firewall have any type of SIP fix-up built in? |
| 18:43.25 | atan | Woah. Okay we're on to something here |
| 18:43.36 | atan | If the ATA adapter calls my DID number it connects and we have audio |
| 18:43.47 | atan | But if they call 7940 I can't hear them |
| 18:44.03 | atan | exten looks like: exten => 7940,1,Dial(SIP/7940,20) ; Temp |
| 18:44.24 | atan | So when they dial 7940 the 7940 rings, I answer, but no audio from them |
| 18:45.03 | *** join/#asterisk jplank (~G_Bove@208-104-67-26.dyn.fttp.comporium.net) |
| 18:45.28 | atan | <PROTECTED> |
| 18:45.32 | lanning | have you tried canreinvite=no? |
| 18:45.44 | jplank | whats the option in sip.conf to keep asterisk from giving out too much information in a failed registration? (i.e. no such extensions, wrong passowrd) |
| 18:46.21 | atan | lanning, canreinvite=no is set on my outbound SIP trunk thinger |
| 18:46.28 | atan | but nowhere else in my sip |
| 18:46.33 | atan | Should I add that to general? |
| 18:46.51 | p3nguin | I would put canreinvite=no in each peer that you don't want to reinvite. |
| 18:47.20 | lanning | try it for the peers you are having issues with. |
| 18:47.25 | p3nguin | jplank: alwaysauthreject = yes |
| 18:48.30 | jplank | ty |
| 18:49.19 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@trir-4d0baeb0.pool.mediaWays.net) |
| 18:49.29 | lanning | canreinvite=no, will force the media to always go through asterisk. |
| 18:52.01 | atan | lanning, canreinvite=no fixed the issue |
| 18:52.24 | atan | p3nguin, or can I just set it globally inside [general] ? |
| 18:52.29 | atan | I set it inside general this time |
| 18:52.34 | atan | I assume this means it applies to everything |
| 18:54.08 | ManxPower | I would set it only for devices that are behind NAT |
| 18:54.23 | ManxPower | setting it for devices on the same network as Asterisk is not useful most of the time. |
| 18:54.39 | ManxPower | for example, set it for your provider's entry in sip.conf, but not for your local phones. |
| 18:54.52 | atan | Asterisk is not running locally |
| 18:54.58 | atan | All the phones are remote |
| 18:55.08 | atan | Asterisk isn't in the same datacenter as the trunk either =\ |
| 18:59.40 | *** join/#asterisk [Outcast] (~anonymous@24-183-177-242.dhcp.oxfr.ma.charter.com) |
| 19:03.48 | atan | Gotta run, be back in a bit |
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| 19:24.19 | *** join/#asterisk [T]ank (~chwall@206.71.78.158) |
| 19:27.33 | Archi_007 | Afternoon, anybody here? |
| 19:28.10 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 19:30.41 | tzafrir | Archi_007, probably |
| 19:32.39 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 19:36.39 | Archi_007 | ah, wow... so many people in here and nobody watching? Afternoon nap? |
| 19:36.48 | Archi_007 | need some ideas on how to figure out why after a while I can't dial out... "All Circuts are busy now" while no calls are going out or coming in. |
| 19:36.57 | Archi_007 | <PROTECTED> |
| 19:51.02 | *** join/#asterisk fofware (~Fabian@host184.190-226-209.telecom.net.ar) |
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| 19:55.50 | jermudgeon | anyone here familiar with 1.6/ethmf/dahdi/redfone? I have sync, but can't get dchan to come up, HDLC errors. Redfone tested good (in loopback, too) on 1.4/zaptel. |
| 20:24.58 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
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| 20:39.43 | *** join/#asterisk saizai (~saizai@unaffiliated/saizai) |
| 20:51.22 | saizai | I am looking to patch Asterisk to comprehend new input tones (e.g. redbox coin tones). Where in the code does it currently determine how to identify given frames? I tried tracking that down, and eventually ended up in some sort of dahdi/narrow/etc split, and... am kinda confused at its crypticness at this point. >< |
| 20:52.01 | saizai | s/narrow/skinny/. Plus h323 and something else |
| 21:03.37 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 21:03.46 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 21:05.52 | ManxPower | saizai, Try #asterisk-dev |
| 21:11.16 | *** join/#asterisk DennisG (~DennisG@ip5454b5b3.adsl-surfen.hetnet.nl) |
| 21:11.27 | sque | in asterisk.conf |
| 21:11.47 | sque | is there any reason for existing a trailing questionmark [directories](!) |
| 21:11.51 | sque | on directories header |
| 21:12.01 | sque | exactly this "[directories](!)" |
| 21:12.12 | sque | ? |
| 21:16.25 | tyman | I'm having problems making inbound calls from the pstn thru my sip provide. I'm just starting with asterisk and i'm almost certain that the problem should be right here in my gist https://gist.github.com/666849 |
| 21:17.25 | tyman | <mynum> and <omitted> had legit info removed for obvious reasons |
| 21:20.23 | tyman | any glaring errors in my configs? |
| 21:21.04 | Khratos | sque: That does not seems like a question mark to me |
| 21:21.09 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:21.12 | Khratos | It seems like a template |
| 21:25.14 | mlsmith9999 | Repost - need some ideas on how to figure out why after a while I can't dial out... "All Circuts are busy now" while no calls are going out or coming in. I even have another SIP provider in the trunk sequence, on the outbound routes, and it still give's me that. Once I issue the reload command via the asterisk console, it's fine, for a while. |
| 21:28.19 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 21:44.40 | jermudgeon | anyone have a mirror for dahdi-linux-complete-2.4.0+2.4.0.tar.gz ? download.asterisk.org is not working for me |
| 21:45.44 | Khratos | mlsmith9999: If you write (on CLI) : soft hangup [and press tab here, twice], do you see any hanged channel? |
| 21:46.05 | Khratos | Like a sombie channel of a supposedly finished call |
| 21:46.21 | WIMPy | It's downloadS.asterisk.org |
| 21:46.23 | Khratos | zombie (sorry, English's not my native language) |
| 21:47.53 | *** join/#asterisk cesar_CR (~cesar@celord.ice.co.cr) |
| 21:49.14 | *** join/#asterisk tootai (~quassel@keewi.tootai.net) |
| 21:51.11 | mlsmith9999 | Khratos: Nope, nothing |
| 21:57.06 | sshock | any way with Park() to have it skip announcing the extension? |
| 21:57.50 | sshock | let's say my dialplan sets PARKINGEXTEN explicitly before calling park, so no need to announce it |
| 21:58.20 | sshock | I looked at the Park() application and I looked in features.conf, but I didn't see anywhere to turn this off |
| 22:00.09 | WIMPy | core show application park |
| 22:00.17 | Khratos | In the moment of the 'All circuits busy', is the provider reachable? |
| 22:00.43 | *** part/#asterisk sshock (~sshock@63.248.133.83) |
| 22:01.05 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 22:01.18 | sshock | WIMPy: I already tried that and there are no params |
| 22:01.24 | sshock | this is * 1.4 btw |
| 22:02.08 | WIMPy | That might be the reason. I see option 's'. |
| 22:02.34 | mlsmith9999 | Khratos: Yes, when I reload, it works again... for a while |
| 22:02.53 | sshock | oh, must be only available in newer * :( |
| 22:04.41 | sshock | anyone played with pygooglevoice? |
| 22:04.59 | WIMPy | Actually on the 1.4 I have access to, there seem to be no options at all. Bad luck. |
| 22:05.09 | sshock | yeah :( |
| 22:05.17 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 22:05.19 | sshock | but I may be able to use Bridge instead; I'm not sure yet... |
| 22:05.32 | sshock | the example that comes with pygooglevoice uses Park and ParkedCall |
| 22:07.02 | WIMPy | AFAIR bridge appeared first in 1.6 as well. |
| 22:07.52 | sshock | hmm, I thought I heard that too, but Bridge does show up in "core show applications" for me |
| 22:08.04 | sshock | and core show application Bridge |
| 22:08.45 | Khratos | mlsmith9999: It's probably something on their side |
| 22:09.02 | Khratos | Could you use a pastebin and show me the block on sip.conf where you have that provider declared? |
| 22:12.34 | sshock | but I haven't figured out yet how the call from GV is going to know the channel to bridge to |
| 22:17.52 | mlsmith9999 | Khratos: http://pastebin.com/N2zJ0xzR |
| 22:19.37 | ManxPower | um, asterisk automatically bridges the two legs of the call. that is why there was never a bridge application before 1.6 |
| 22:20.59 | p3nguin | sshock: Why would you want it to not tell you where the call is parked? That's how you know how to pick up the call later. |
| 22:21.52 | *** join/#asterisk n0tk (~n0tk@216.160.42.30) |
| 22:21.52 | sshock | p3nguin: this is a scenario where I explicitly set PARKINGEXTEN before calling Park() |
| 22:22.32 | sshock | I'm playing with pygooglevoice, but I'll use Bridge instead of Park and ParkedCall if I can... |
| 22:22.44 | pabelanger | mlsmith9999: what version of Asterisk? |
| 22:23.39 | sshock | hmm, looks like I could save the current channel to the * DB before doing the System(gvoice call) |
| 22:23.44 | pabelanger | mlsmith9999: and how long does it work before asterisk stop processing outbound calls? |
| 22:23.52 | sshock | then when the call comes in, I can look that up and bridge to it |
| 22:24.44 | mlsmith9999 | pabelanger: 1.4 |
| 22:25.12 | pabelanger | mlsmith9999: 1.4.what? |
| 22:25.28 | mlsmith9999 | pabelanger: anywhere between 1-2 hours. |
| 22:25.59 | mlsmith9999 | pabelanger: sorry, 1.4.22-4 |
| 22:26.37 | pabelanger | mlsmith9999: Hmm, I wonder if it is a dnsmgr issue. |
| 22:26.40 | pabelanger | ~collectdebug |
| 22:26.41 | infobot | [collectdebug] a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 22:27.09 | pabelanger | mlsmith9999: ^^ try and capture a debug log next time it happens |
| 22:27.22 | mlsmith9999 | ok. |
| 22:29.30 | mlsmith9999 | pabelanger: it say's in the Prerequisites * Asterisk 1.4.30 or greater. |
| 22:30.01 | pabelanger | mlsmith9999: you should be fine with 1.4.22-4, I just tested the commands with 1.4.30 when I wrote the document |
| 22:31.25 | mlsmith9999 | oh, ok... will do... |
| 22:32.43 | *** join/#asterisk qubix (qubix@unaffiliated/qubix) |
| 22:32.43 | ManxPower | you can't usually collect debug from Asterisk installed via a package manager |
| 22:32.44 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 22:33.03 | ManxPower | the -4 usually means "packaged version" |
| 22:34.00 | pabelanger | ManxPower: Sure you can; backtrace's however are a different story |
| 22:34.59 | pabelanger | mlsmith9999: ManxPower raises a good point, can you reproduce this with a version of asterisk download and compiled from asterisk.org ? |
| 22:35.49 | atan | Anyone here know the key differences between the polycom 500 and 550? |
| 22:36.04 | qubix | different numbers |
| 22:37.52 | atan | I'm just confused why the 550 is so much more than the 500. |
| 22:38.01 | qubix | 50's a higher number. |
| 22:39.42 | qubix | I was going to compare on the polycom website for you; but the 500 is no longer on there, so I can't... |
| 22:41.16 | ManxPower | BTW, the asterisk rpms from asterisk .org have no documentation included |
| 22:41.31 | ManxPower | atan, the 500's were discontinued several years ago |
| 22:41.47 | ManxPower | none of the recent firmware runs on them |
| 22:42.22 | atan | I see. |
| 22:45.16 | atan | Have either of you used SIP video before? |
| 22:45.28 | qubix | I'm too ugly for video phones |
| 22:45.43 | atan | Is it hit-or-miss? Are all the "SIP Video" phones compatible with one another or would one need all the same brands of phones? |
| 22:45.55 | atan | I've notices several suppliers selling video SIP phones =\ |
| 22:48.13 | robl^laptop | the 550 is MUCH advanced over the 500. HD voice (g722), faster processor, better display, backlit, more memory, etc |
| 22:48.20 | mlsmith9999 | pabelanger: not right now.. This thing is in production... don't ask we had to move fast our prior VoIP solution was being shutdown. Buyout. So used the opportunity to move to asterisk. |
| 22:49.57 | mlsmith9999 | BTW, does anybodu here have Cisco SPA504G phones working with xml files and SLA? |
| 22:50.03 | mlsmith9999 | *anybody |
| 22:51.05 | ManxPower | I dont' do video, I have no interest in doing video |
| 22:52.10 | pabelanger | mlsmith9999: http://en.wikipedia.org/wiki/Project_triangle :) |
| 22:53.34 | mlsmith9999 | pabelanger: ah, yeah nice.. well how about never again this proprietary $#%@ and I'm talking about Linksys One. |
| 22:54.10 | mlsmith9999 | Now I have 2 SVR3000 and 7 VoIP phones I can't use. |
| 22:54.30 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 22:57.10 | *** part/#asterisk sshock (~sshock@63.248.133.83) |
| 22:59.21 | mlsmith9999 | pabelanger: got the log, you want me to pastebin it? |
| 23:02.48 | atan | ManxPower, probally for the better... I can't find any solid video standard =S |
| 23:24.24 | p3nguin | mlsmith9999: What's wrong with Linksys? |
| 23:25.45 | mlsmith9999 | p3nguin: End of Life for the LinksysOne platform. Cisco bout Linksys and decided that they already have a small business VoIP platform and EOL'ed LinkSysOne. We had it for 3 Years. |
| 23:26.37 | *** join/#asterisk jermudgeon (~jhaustin@216-67-61-242.static.acsalaska.net) |
| 23:27.48 | mlsmith9999 | *bought |
| 23:28.58 | mlsmith9999 | pabelange: were you wanting me to post that as a bug, or for you to look at? |
| 23:29.40 | p3nguin | mlsmith9999: What's the issue with an Asterisk migration? You've got the necessary hardware, right? |
| 23:31.09 | mlsmith9999 | p3nguin: Yeah had to buy all new PBX and Phones (cisco SPA504G) The LinksysOne Hardware is proprietary to their service. |
| 23:31.49 | p3nguin | mlsmith9999: If you bought a new PBX, what do you intend to do with Asterisk? |
| 23:32.21 | mlsmith9999 | p3nguin: Well the pbx is actually an asterisk machine. |
| 23:33.32 | p3nguin | mlsmith9999: But you bought a new set of SPA504G phones and built an Asterisk system? I'm still trying to find out what the actual problem is, but you aren't helping me out. |
| 23:35.11 | mlsmith9999 | p3nguin: Oh, ok sorry you missed the prior conversation. I have an outgoing trunk that about every hour or so stops allowing outgoing calls. I get all circut's are busy now. If I issue the reload command at the CLI, it works again. |
| 23:35.59 | mlsmith9999 | p3nguin: I posted the debug log for pabelange. he asked for it. |
| 23:35.59 | p3nguin | mlsmith9999: An ITSP? Which one? |
| 23:36.15 | *** join/#asterisk sbszulu (~dundubala@41.14.47.118) |
| 23:36.15 | mlsmith9999 | p3nguin: Velocity |
| 23:37.11 | mlsmith9999 | p3nguin: we had to move the box from behind the firewall and dualhome it just to get it to work the way it is now. |
| 23:38.04 | p3nguin | mlsmith9999: Have you tested any others to see if the same thing happens? I'd be concerned about a not-so-well-known ITSP having quality services. |
| 23:40.06 | mlsmith9999 | p3nguin: Yes I've tried it with vitelity but they can't port our number to them. plus we already have a three year conrtact to keep our former rates. And yes it worked with Vitelity. They were my first choice since I've had an account with them for 5 Years now, without a hitch. |
| 23:40.18 | mlsmith9999 | *contract |
| 23:41.50 | p3nguin | mlsmith9999: So you already know the problem is with this no-name company, but you're trying to "fix" your equipment, which apparently isn't broken? I don't get it. |
| 23:42.06 | *** join/#asterisk nicoAMG (~nicoAMG@201.237.49.131) |
| 23:42.56 | mlsmith9999 | p3nguin: We don't know it's with them, that's why we're debugging it. |
| 23:43.34 | *** join/#asterisk calmh (~jb@acro.nym.se) |
| 23:43.53 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 23:44.04 | mlsmith9999 | p3nguin: And this"no-name" company is an off shoot of Commpartners which is actually large. |
| 23:44.42 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 23:44.45 | mlsmith9999 | p3nguin: plus that's part of why I'm here, is, if it is thier problem, to prove it. |
| 23:44.55 | mlsmith9999 | *their |
| 23:45.19 | mlsmith9999 | German/American mixing into each other there... |
| 23:49.14 | tyman | repost: I'm having problems making inbound calls from the pstn thru my sip provide. I'm just starting with asterisk and i'm almost certain that the problem should be right here in my gist https://gist.github.com/666849 |
| 23:49.16 | *** join/#asterisk dev_ast (~chatzilla@cpe-76-182-207-42.tx.res.rr.com) |
| 23:49.33 | tyman | <mynum> and <omitted> had legit info removed for obvious reasons |
| 23:50.13 | dev_ast | Has anyone successfully implemented Shared Line Appearance with Asterisk and Polycom 335 phones |
| 23:52.39 | mlsmith9999 | or Asterisk and Cisco SP504G? |
| 23:57.37 | mlsmith9999 | sorry tyman but I'm not good enough with asterisk to help you. |
| 23:58.11 | tyman | mismith9999: np, thx |