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| 01:33.22 | przerull | on asterisk 1.8 ${HASH(SIP_CAUSE,${CDR(dstchannel)})} isn't returning anything after a Dial which returned DIALSTATUS=CONGESTION. Is there a way to get the sip response code from the dial? |
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| 03:06.49 | hebber | Any advice on FXO 4 ports Gateways? |
| 03:07.21 | hebber | I have problems with Grandstream GXW 4104 |
| 03:10.33 | igcewieling | I don't know of anything with close to that small number of ports |
| 03:11.16 | hebber | what about FXO Gateways in general? |
| 03:16.00 | Kobaz | hebber: cisco |
| 03:16.15 | Kobaz | the grandstreams are awful |
| 03:16.45 | Kobaz | and for something better than the cisco, go with audiocodes |
| 03:16.50 | Kobaz | they make a really nice 4 port |
| 03:17.12 | Kobaz | mp114-fxo |
| 03:23.09 | igcewieling | We use Adtran Netvanta media gateways |
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| 03:23.33 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 03:26.56 | hebber | kobaz: thanks for the tip - did you figure out the missing ACK problem? |
| 03:27.34 | Kobaz | i think so |
| 03:27.42 | Kobaz | i'll see tomorrow |
| 03:28.11 | Kobaz | i had swapped the order of an unlock/free |
| 03:28.20 | Kobaz | so i had code writing to free'd memory |
| 03:28.45 | Kobaz | which can cause all kinds of problems, basically undefined behavior |
| 03:29.32 | Kobaz | i had another site that was getting random call hangups |
| 03:33.26 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 03:33.38 | Kobaz | it was the result of a fix for a deadlock |
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| 03:40.00 | hebber | hmm - that was over my head |
| 03:41.59 | Kobaz | heh, aw |
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| 03:51.56 | Kobaz | basically my code was stepping on memory it shouldn't have been |
| 03:52.45 | Kobaz | and all kinds of problems can result from that since at that point there's no telling what was overwritten and what will use it |
| 03:55.13 | carrar | roulette malloc? |
| 03:55.58 | carrar | no one can ever run it seven times in a row |
| 03:56.37 | hebber | kobaz: I see, I just makes scripts - I'm no programmer |
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| 04:04.00 | Kobaz | carrar: free(foo); AST_RWLIST_UNLOCK(foo) |
| 04:04.12 | Kobaz | == bad |
| 04:04.20 | Kobaz | i found it with valgrind |
| 04:05.16 | Kobaz | it would be "fine" in a single thread application, but with asterisk and who knows 50-100 threads going on, there's no telling what will execute in between the free and the unlock |
| 04:11.40 | igcewieling | When asking for help here it is usually a good idea to nor be running with your own patches. |
| 04:13.39 | Kobaz | well i didn't know it was my own code causing trouble at first |
| 04:14.39 | Kobaz | you'll never see me asking for help in here if my own stuff is blowing up |
| 04:14.49 | Kobaz | (if known) |
| 04:16.02 | Kobaz | but i can't not run my own branch... everything built depends on it |
| 04:16.04 | Kobaz | but anyway |
| 04:16.07 | Kobaz | nap time |
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| 04:49.53 | atan | Is there something nifty built into Asterisk that would allow you to dictate? I see some stuff on voip-info.org about cmd Distate but wasn't sure if it was still around. Looking at http://www.asterisk.org/get-started/features I don't see it listed |
| 04:54.47 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:04.32 | kaldemar | atan: check your version for the apps. but yes, dictate still exists in asterisk. |
| 05:09.38 | atan | kaldemar, I'll poke around in menuselect for it. |
| 05:10.46 | atan | I just jumped from 10.8 to 11.3 and something broke on me... go figure eh? Now to figure that out and then play with dictate. Right now it's moaning about not finding a col named callbackextension but I don't request that anywhere in my odbc query, searching for details about that |
| 05:12.05 | atan | found app_dictate btw, thanks :-) glad to find it |
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| 05:17.51 | *** join/#asterisk frek818 (~herman@cpe-76-169-198-140.socal.res.rr.com) |
| 05:18.31 | frek818 | Hi, can anyone point me to the must up to date information about Asterisk and virtualization? |
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| 05:21.21 | frek818 | To make myself more clear: Can anyone point me to the most up to date information about Asterisk and virtualization/containerization. Specifically, I would like to figure out which route is the best. I have an idea about how important timing is. |
| 05:22.04 | ChannelZ | It's slightly less important than it used to be |
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| 05:32.37 | frek818 | ChannelZ: Thanks |
| 05:33.07 | frek818 | I'll get some experiments going. |
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| 05:42.18 | ChannelZ | Zaptel/DAHDI timing used to be necessary for MeetMe conferences and something else I'm forgetting (MixMonitor records?) and was pretty sensitive. |
| 05:45.21 | ChannelZ | Nowadays there's a couple different alternate methods (timerfd, pthreads) that are probably more tollerable in vm environments. I mean lots of people are running Asterisk virtualized, but that said, I don't have any personal experience or suggestions |
| 05:47.57 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.106) |
| 05:55.51 | atan | How do you force the dialplan to continue despite a caller hanging up? |
| 05:59.12 | atan | looks at this 'h' option |
| 06:00.16 | iprouteth0 | http://www.voip-info.org/wiki/view/Asterisk+h+extension |
| 06:01.20 | *** join/#asterisk k611 (~K610@cable-78.29.241.186.coditel.net) |
| 06:02.34 | atan | I have same => h,n,System( in there but it's not getting called |
| 06:03.40 | iprouteth0 | hmmm... I've never used it myself. could you pastebin that chunk of dialplan? |
| 06:04.04 | atan | You bet. Thanks for the help, just one sec |
| 06:04.40 | atan | http://pastebin.com/gtzuZ8Ns |
| 06:05.59 | iprouteth0 | try it without the hangup statement |
| 06:06.52 | atan | Removed hangup() and it still doesn't get hit, |
| 06:06.59 | atan | Watching in debug shows it's not being run sadly |
| 06:07.31 | iprouteth0 | and take the n sequence out since the h replaces it |
| 06:08.04 | atan | pbx_config.c:1586 pbx_load_config: Invalid priority/label 'h' at line |
| 06:08.05 | iprouteth0 | same => h,System(command) |
| 06:08.47 | atan | Yeah that's what I have in there now, results in that error on reload (same => h,System() |
| 06:08.52 | iprouteth0 | or I might be wrong. it might be |
| 06:09.25 | iprouteth0 | exten => h,n,System(command) |
| 06:09.30 | iprouteth0 | instead of |
| 06:09.46 | iprouteth0 | exten => same,h,System |
| 06:10.02 | atan | ...hmmmmmm, let's try that |
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| 06:11.24 | kaldemar | atan: h is a separate extension. it must start with priority 1. you had n there. |
| 06:11.53 | iprouteth0 | ok, that makes sense |
| 06:12.34 | iprouteth0 | so.. |
| 06:12.45 | iprouteth0 | exten => h,1,System(command) |
| 06:13.23 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.156) |
| 06:13.28 | atan | Okay here's where I am now, http://pastebin.com/WjBuRpjh (reading comment from kaldemar and iprouteth0 now) |
| 06:13.55 | atan | Won't that throw an error saying 1234 already has line #1? |
| 06:14.01 | kaldemar | ... |
| 06:14.13 | kaldemar | 1,h != h,1 |
| 06:14.36 | kaldemar | also it is not the same exten, so don't use "same". |
| 06:14.59 | atan | Ohhhhhh so I'm going to need to make a whole new context for this thing |
| 06:15.09 | kaldemar | atan: NO. |
| 06:15.19 | kaldemar | atan: extension, not a context. |
| 06:15.22 | iprouteth0 | no just put your statement at the end |
| 06:15.32 | iprouteth0 | so |
| 06:15.35 | iprouteth0 | exten => _1NXXNXXXXXX,n,Dial(Motif/iprouteth0/${EXTEN}@voice.google.com,,r) |
| 06:15.50 | iprouteth0 | exten => h,1,System(command) |
| 06:17.07 | kaldemar | whether it is at the end or the beginning is irrelevant, as long as it is in the "current context", i.e. the one which was last went to with goto or dial (or which was set as a context in a channel configuration file). |
| 06:17.27 | atan | Ahuh! Success! |
| 06:17.34 | atan | You guys rock so much :-) |
| 06:17.40 | atan | Thank you both |
| 06:17.49 | iprouteth0 | him more than me, lol |
| 06:18.12 | atan | I'm still happy you were both around ;) thank you |
| 06:18.22 | iprouteth0 | I have only half decent understanding of dialplan code |
| 06:18.56 | iprouteth0 | always happy to help though |
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| 06:32.17 | atan | Hey iprouteth0 and kaldemar, if I wanted to run two System() commands how would I start the second, third line below the first exten => h,1,System(? For some reason I thought exten => n,System( but that doesn't appear to be the case |
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| 06:33.09 | kaldemar | atan: you seriously need to learn what extensions and priorities are and their syntax. |
| 06:33.18 | iprouteth0 | exten => h,1,System(command1) |
| 06:33.29 | iprouteth0 | exten => h,n,System(command2) |
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| 06:33.52 | kaldemar | atan: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html |
| 06:33.59 | iprouteth0 | n is always a priority not an extension |
| 06:34.12 | iprouteth0 | s is the one that is sometimes used as an extension |
| 06:34.57 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions |
| 06:35.27 | atan | hits self with large rock. |
| 06:35.41 | atan | I had an 'n' in where I thought I had an 'h'... back to primary I go :-) |
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| 06:45.58 | *** join/#asterisk bulkorok (~chatzilla@85.183.36.36) |
| 06:46.35 | bulkorok | hi |
| 06:47.00 | iprouteth0 | hello |
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| 07:04.30 | *** join/#asterisk BeeBuu (3afc49a4@gateway/web/freenode/ip.58.252.73.164) |
| 07:04.45 | BeeBuu | hi,all. |
| 07:05.03 | atan | This .raw file that Dictate leaves me with, what exactly is that? I am looking around on the first few pages of Google results for people using sox to convert this into a wav but many of the commands I've found have args that aren't in my version of sox (-w for example) |
| 07:05.09 | atan | Hello BeeBuu |
| 07:05.18 | BeeBuu | how many userfield i can add to CDR ? |
| 07:05.37 | BeeBuu | and how to add more userfield ? |
| 07:08.34 | atan | Hmm okay so I guess -w is now just -b 16, guess that's solved |
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| 07:11.12 | kaldemar | BeeBuu: userfield is just userfield. you can add as many fields as you like but then you need cdr_custom or similar to actually write them to cdr. |
| 07:12.34 | BeeBuu | kaldemar: thanks.any URL about that? |
| 07:13.52 | bulkorok | BeeBuu: https://wiki.asterisk.org/wiki/display/AST/CDR+Variables |
| 07:13.59 | bulkorok | last sentence |
| 07:15.09 | bulkorok | additionally: Using MySQL for CDR records is supported by using ODBC and the cdr_adaptive_odbc module (depends on res_odbc). |
| 07:15.44 | BeeBuu | just so simple? CDR(name)? |
| 07:16.27 | bulkorok | seems like :-) |
| 07:16.28 | BeeBuu | thanks bulkorok & kaldemar |
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| 07:39.08 | uosiu | Hi all |
| 07:41.03 | uosiu | I have a problem with my Asterisk 1.8.20.1 on debian squeeze |
| 07:41.49 | uosiu | I have executed "core set debug 99" and "core set verbose 99" to check did incoming calls over SIP are coming to my PBX |
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| 07:45.09 | uosiu | sip show peers shows that my upstream provider is OK, local clients also |
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| 07:52.20 | hebber | uosiu: what is the problem? |
| 07:52.40 | uosiu | hebber: no incoming calls to my PBX |
| 07:53.02 | bulkorok | uosiu: sip set debug on and check for INVITEs |
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| 09:57.18 | linocisco | hi all |
| 09:58.22 | linocisco | i want users to press 91 and wait for dial tone and if only line is free , input desired no. . if line 1 is not free, pressing 91 should proceed to next available line automatically . how could I do that? |
| 09:58.35 | *** join/#asterisk _omer (~omer@116.71.178.187) |
| 09:59.29 | _omer | using Asterisk 11 Realtime .... when I register and sip user, I get following message on Asterisk CLI .... [Apr 26 05:53:24] WARNING[3313]: chan_sip.c:30571 build_peer: '' is not a valid RTP hold time at line 0. Using default. |
| 09:59.33 | _omer | any help? |
| 10:02.00 | kaldemar | what do you want to do with that? |
| 10:02.39 | kaldemar | linocisco: what do you mean by line? |
| 10:03.21 | _omer | [Apr 26 05:53:14] WARNING[3313]: chan_sip.c:30571 build_peer: '' is not a valid RTP hold time at line 0. Using default. |
| 10:03.22 | _omer | [Apr 26 05:53:14] WARNING[3313]: chan_sip.c:30420 build_peer: no value given for outbound proxy on line 0 of sip.conf |
| 10:03.25 | linocisco | kaldemar, PSTN lines which are connected to grandstream GXW4104 FXO gateways |
| 10:03.31 | _omer | this is what I see on Asterisk CLI ... |
| 10:08.28 | kaldemar | _omer: and what do you want to do about those lines? |
| 10:09.05 | kaldemar | _omer: they are just warnings about you not having optional parameters in your database. if you don't want to see those, set some values for them. |
| 10:11.19 | kaldemar | linocisco: how do you differentiate using those lines on the asterisk side? |
| 10:12.41 | linocisco | kaldemar, what do u mean? |
| 10:13.45 | kaldemar | linocisco: how do you dial line 1? how do you dial line 2? |
| 10:15.20 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 10:15.37 | linocisco | kaldemar, i want to press 91 and when I hear dial tone which means that line is free, i will put destination no. 242828 which is land line in the same city, (074)50010 which is calling to another city, 0943127800 which is for mobile |
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| 10:30.21 | kaldemar | linocisco: yes, i understand what you want to do. how do you dial from asterisk using line 1 of the gateway? |
| 10:31.34 | linocisco | kaldemar, i press 91 and dial to hear dial tone. no way. "your call can'tbe completed bla bla" |
| 10:34.20 | kaldemar | "from asterisk" meaning what does your Dial() look like? what you press in your phone is irrelevant. |
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| 10:41.36 | linocisco | kaldemar, so what do i do? |
| 10:41.43 | *** join/#asterisk sjs205 (~sjs205@host81-155-250-128.range81-155.btcentralplus.com) |
| 10:41.47 | sjs205 | Morning all |
| 10:42.41 | cusco | http://1.bp.blogspot.com/-UFaBPD5a7F8/UVBSDO4XlPI/AAAAAAAACs0/iyk6cFCQ3lw/s1600/monday-motivate-16.jpg |
| 10:42.44 | cusco | oops |
| 10:42.46 | cusco | sorry |
| 10:42.47 | cusco | I meant: hi |
| 10:59.04 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 11:00.19 | sjs205 | I'm looking at implementing a system whereby a small c app initiates 2 calls, one of them will just be a MOH, the other will provide the and IVR based system, and based on the recipients input, should merge the two calls. Any ideas on which method would be best to implement such a system? |
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| 11:05.41 | WIMPy | linocisco: Didn't we discuss that last week? What happened since then? |
| 11:06.01 | linocisco | WIMPy, stil not ok to call outside |
| 11:06.34 | *** join/#asterisk evilsk4ter (~evilsk4te@187.60.66.11) |
| 11:06.55 | WIMPy | Oh, I see the lines are not connected to Asterisk. that's bad. |
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| 11:33.43 | vlad_sta_ | Question: Anyone one had an issue with no outgoing calls on Grandstream DP715? |
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| 12:30.47 | Onyx47 | Hello, I have a discrepancy between two server: on both I use CDR(src) in Queue's GoSub argument, on one of them I can see it fine, on the other one it shows the number of extension that answered the call instead of the one that entered the queue and was answered. Can anyone confirm which behaviour is normal? |
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| 12:32.51 | [sr] | hi WIMPy |
| 12:33.01 | vlad_sta_ | Question: Anyone with Grandstream phones knowledge here? I have an issue with outgoing calls on DP715 with TLS enabled. Need help. |
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| 12:46.09 | danfromuk | Does anyone know how to secure polycom phones to ensure they only accept sip calls coming from the asterisk server? |
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| 13:03.15 | twanny796 | is there astlog for windows? |
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| 13:04.28 | krotos | Hi all chat |
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| 13:04.55 | krotos | quick question, i'm start playing around WebRTC socket support, for make videocall |
| 13:05.09 | krotos | is videocall supported with asterisk 11, and WebRTC? |
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| 13:18.22 | subthalamus | anybody willing to try to call my sip? |
| 13:18.36 | subthalamus | or is there a test facility to get autocalled? |
| 13:19.24 | jmetro | dial yourself? |
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| 13:20.47 | subthalamus | jmetro: urm, wouldn t that just give me a busy tone or voicemail? |
| 13:21.50 | jmetro | from a different phone, like a softphone on your pc |
| 13:22.13 | subthalamus | I'm receiving on a softphone on my pc |
| 13:22.21 | jmetro | softphone on pc#2? |
| 13:23.47 | subthalamus | jmetro: I think the ISP (sip provider) is doing something wonky |
| 13:24.05 | subthalamus | nothing works |
| 13:24.22 | jmetro | be more verbose |
| 13:25.31 | subthalamus | ok, problem 1: I'm not well versed in voip 2: we've never gotten this to work except for incoming calls to the normal sip number |
| 13:25.52 | subthalamus | aka not through a sip gateway for example |
| 13:26.19 | subthalamus | problem 3: the isp provides 2 settings, nothing more and no documentation |
| 13:27.04 | subthalamus | I think until I figure out this end there is no point in talking about the INUM DID forwarding issues |
| 13:27.57 | jmetro | right |
| 13:28.01 | jmetro | what do they provide you |
| 13:28.42 | subthalamus | these r the 2 settings Redirection appels entrants : vers mon téléphone SIP X vers mon téléphone Freebox X |
| 13:29.06 | subthalamus | translation: forward to SIP phone or forward to freebox phone |
| 13:29.34 | subthalamus | the freebox is a modem/router into which normal analog phones are plugged |
| 13:29.47 | jmetro | it sounds like they are providing you a single endpoint |
| 13:29.51 | subthalamus | rj25 port or whatever |
| 13:30.37 | jmetro | so your problem is that you can only see your calls coming in if you dial the sip number |
| 13:30.51 | jmetro | the Sip URI |
| 13:31.22 | subthalamus | um, the phones plugged into the freebox ring only if the regular number is called from a regular phone |
| 13:31.53 | subthalamus | regardless of which setting is selected |
| 13:32.10 | jmetro | do you know what your Sip URI should be? |
| 13:32.15 | subthalamus | any attempt to call number@freephonie.net fails |
| 13:32.38 | subthalamus | yes, <our number>@freephonie.net |
| 13:33.04 | jmetro | ok so your "PSTN" number works but your SIP URI doesnt |
| 13:33.20 | subthalamus | correct |
| 13:35.09 | jmetro | where do you have asterisk? |
| 13:35.28 | subthalamus | nowhere |
| 13:36.22 | jmetro | i woulddnt be able to see what is wrong if I dialed it, that would all be up to them and their log files.. i'd suggest contacting them |
| 13:36.23 | subthalamus | is this a facepalm moment? |
| 13:37.15 | jmetro | how exactly does dialing the URI fail? What happens? just busy? |
| 13:37.23 | subthalamus | so I could use callcentric or something else if I dedicated a pbx box here at home? |
| 13:38.22 | jmetro | i wouldnt know about that, my situation is SIP Trunk provider with DID's, -> my asterisk box -> phones |
| 13:38.33 | subthalamus | well, u mean if I dial my own softphone from my own softphone? |
| 13:39.01 | jmetro | put another softphone on a different PC [you cant use vmware or virtualboxes for softphones] and dial it |
| 13:39.09 | subthalamus | what is strange is the softphone seems to login fine and I can make outgoing calls |
| 13:39.31 | jmetro | it is something with their routing probably, which is why i would suggest talking to freephonie |
| 13:40.14 | subthalamus | hmm, ok, thanks for the sanity check I guess :) |
| 13:40.50 | jmetro | you could also stick asterisk on something and register a phone to it to make a sip URI call =) |
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| 13:41.51 | subthalamus | I don't exactly have computers coming out of my ears sadly |
| 13:42.02 | subthalamus | I'll figure something out |
| 13:42.08 | jmetro | in this case you could use a virtual box |
| 13:42.32 | jmetro | download debian and put it on there, install asterisk, and register over the network |
| 13:42.42 | subthalamus | heh, these rickity things can barely run the os with 1GB of ram |
| 13:43.00 | jmetro | youd be surprised, debian server can run on a very low end pc |
| 13:43.20 | subthalamus | yea, I know, guess I'll bite the bullet |
| 13:44.15 | jmetro | do it virtually so you dont have to erase the thing |
| 13:44.21 | jmetro | vmware player is also free |
| 13:44.51 | jmetro | plus afterwards you have a working asterisk box you can play with =) |
| 13:49.21 | subthalamus | is there a reason you're recommending vmware over virtualbox? |
| 13:50.36 | jmetro | I've used both for just small-scale desktop virtual machines, and VMware player feels more like a real machine, less lag. Theyre virtually identical technically though. |
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| 14:21.05 | Rahoul | hi I am using an Avaya phone with Asterisk and the problem I have is that I cannot dial into any of the feature codes. Whenever I dial it from the Avaya phone I get the following on the sip debug log |
| 14:21.19 | jmetro | please use a pastebin for big logs |
| 14:22.25 | Rahoul | http://pastebin.com/84pvbg73 |
| 14:22.29 | Rahoul | sure |
| 14:23.28 | Rahoul | if I dial 918xxxxxxxxx it passes it at 918xxxxxxxxx@my.domain.com but for example voicemail (*97) passes it only as <*97> without the @my.domain.com |
| 14:23.42 | Rahoul | as shown on the log |
| 14:23.58 | Qwell | Disable any sort of feature code support on the phone. |
| 14:24.16 | Rahoul | this is the dial plan setting I am using on the phone |
| 14:24.24 | Qwell | refer to Avaya docs |
| 14:24.24 | Rahoul | SET DIALPLAN [23]xxxx|91xxxxxxxxxx|9[2-9]xxxxxxxxx|*xx |
| 14:24.38 | Rahoul | @Qwell trust me I did... |
| 14:24.42 | Rahoul | they suck... |
| 14:24.57 | Qwell | get a new phone? |
| 14:25.11 | Rahoul | well I bought this baby off ebay (used) |
| 14:25.24 | Rahoul | I do have better phones to use but I wanted to get this one up and running |
| 14:25.29 | Qwell | sucks. buy a new phone? |
| 14:25.38 | Rahoul | yeah could be... |
| 14:25.54 | Rahoul | but i would like to play around so that I can get access to asterisk feature codes |
| 14:26.00 | Rahoul | b/c the rest is working great |
| 14:26.06 | Rahoul | :( |
| 14:27.42 | igcewieling | Rahoul: what exactly is not working. I don't mean the sip messages, I mean in Asterisk |
| 14:28.32 | Rahoul | its not asterisk, its this phone set. It allows me to make calls but I am unable to call any *xx through it. |
| 14:28.41 | Rahoul | for some reason it's not passing them correctly to asterisk |
| 14:29.10 | jmetro | wouldnt your phone just dial "*97" and then your local routing dialplan for asterisk would pick up *97 and translate it properly |
| 14:29.29 | jmetro | its like dialing "100" you dont dial "100@domain" you just dial "100" and dialplan does the rest |
| 14:31.00 | Rahoul | exactly! |
| 14:31.07 | igcewieling | Rahoul: sip debug on Asterisk shows the call not arriving on Asterisk? |
| 14:31.18 | Rahoul | if I dial 100 from the phone, asterisk gets 100@domain |
| 14:31.27 | Rahoul | but if I dial *97 asterisk only gets *97 |
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| 14:32.11 | igcewieling | Rahoul: Do you have an exten => *97 in your dialplan? |
| 14:32.18 | Rahoul | yes |
| 14:32.27 | Rahoul | all feature codes work great with a softphone |
| 14:32.32 | Rahoul | or with a cisco ip phone |
| 14:32.38 | Rahoul | but with avaya I get the following |
| 14:32.50 | Rahoul | for 100 for example <sip:100@pbx.mydomain.com> |
| 14:33.19 | Rahoul | but for *43 i get <sip:*43> |
| 14:33.58 | Rahoul | the phone is not realising it should append @pbx.mydomain.com to *xx |
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| 14:40.38 | Rahoul | any ideas? |
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| 14:44.23 | jmetro | add the text you need to the feature on the phone |
| 14:47.07 | igcewieling | ADTRAN, Inc. OS version R10.6.0.E <-- written by satan himself |
| 14:49.25 | jmetro | i like netgear prosafe stuff |
| 14:49.36 | Rahoul | so for example my dial plan should be: SET DIALPLAN [23]xxxx|91xxxxxxxxxx|9[2-9]xxxxxxxxx|*xx@my.domain.com |
| 14:49.51 | jmetro | try it out, why not. |
| 14:49.57 | Rahoul | hang on |
| 14:50.00 | Rahoul | good idea |
| 14:50.15 | jmetro | if it doesnt wanna work the way you want it to, break it to make it work the way it should |
| 14:54.46 | Rahoul | jmestro: please take a look at this http://pastebin.com/LajX8FMC |
| 14:56.19 | igcewieling | Rahoul: what does Asterisk do when it receives the *43 without the @xxx? |
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| 14:56.58 | Rahoul | echo test |
| 14:57.05 | Rahoul | (using freepbx) |
| 14:57.31 | Rahoul | if i call *43 from ekiga it is passed correctly with the @my.domain.com |
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| 14:58.55 | Rahoul | and the echo test starts |
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| 15:01.56 | jmetro | <PROTECTED> |
| 15:02.36 | Rahoul | as shown in the pastebin link it treats it as Unauthorized |
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| 15:02.49 | Rahoul | which makes sense, because asterisk does not know what domain is *43 into |
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| 15:03.23 | jmetro | the phone should have a default context it goes into |
| 15:06.07 | Rahoul | i think it uses the right domain for all calls but those that start with * |
| 15:06.39 | Rahoul | for example in order to call 1800 555 6666 the phone correctly passes 918005556666@my.domain.com |
| 15:06.49 | Rahoul | however it does not do the same thing for *xx |
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| 15:18.57 | igcewieling | Rahoul: Again, the issue is the phone, not Asterisk |
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| 15:19.34 | Rahoul | i got it to work using URI dialing |
| 15:19.42 | Rahoul | now I need to define speed dials as URI |
| 15:19.46 | Rahoul | URIs* |
| 15:19.49 | Rahoul | and it will work :) |
| 15:20.22 | jmetro | are avayas [nortels] easy to get on Asterisk? ive never bothered with them since polycoms/aastras/ciscos are so much nicer |
| 15:21.21 | Rahoul | Im a linksys/cisco/softphone fan |
| 15:21.36 | Rahoul | however this avaya 4621 was a bargain! |
| 15:21.39 | Rahoul | I had to try.. |
| 15:22.01 | hexanol | I have a problem with app_queue |
| 15:22.09 | hexanol | I have the following setup: 2 queues, with both the same available member (i.e. same interface, state_interface, membername) |
| 15:22.12 | Rahoul | besides its always nice trying to figure out how to make other devices work |
| 15:22.15 | hexanol | If I call both queues simultaneously |
| 15:22.19 | hexanol | Then my member's interface is called twice |
| 15:22.27 | jmetro | ringinuse=no ? |
| 15:22.45 | hexanol | ringuse is disabled |
| 15:22.57 | hexanol | there's a SIP/.. state_interface |
| 15:23.06 | hexanol | *these are |
| 15:23.19 | hexanol | in fact, if I call with 500 ms apart |
| 15:23.20 | hexanol | it's fine |
| 15:23.28 | hexanol | but if I call both queues exactly at the same time |
| 15:23.30 | hexanol | it's not |
| 15:24.20 | hexanol | on asterisk 11.3.0, but I've also tested on asterisk 1.8, same behaviour |
| 15:24.29 | jmetro | hm. i guess it comes down to lucky people hitting the dial button at the same time |
| 15:25.07 | hexanol | well, this a simple way to reproduce the problem in a test environment |
| 15:26.15 | hexanol | in fact, if one call enter the queue at T0, and one call enter at exactly T0 + ~1000 * X ms, then the bug can also happens |
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| 15:27.37 | Greenlight | hexanol: This is a known bug |
| 15:28.01 | hexanol | Greenlight: it is documented somewhere / is there a ticket on the asterisk tracker ? |
| 15:28.07 | hexanol | *is it |
| 15:28.32 | Greenlight | Asterisk-16115 i beleive |
| 15:28.33 | LieutPants | [ASTERISK-16115] [Status: Reopened] [Assigned: marquis] [patch] problem with ringinuse=no, queue members receive sometimes two calls - https://issues.asterisk.org/jira/browse/ASTERISK-16115 |
| 15:28.43 | Greenlight | That sound like what you're seeing ? |
| 15:29.21 | hexanol | yes |
| 15:29.47 | Greenlight | Don't expect a fast resolutionk; that ones beenm open a while. I think most folks have worked around it somehow or other (I had to). I think the crux of the issue is how the queues are handled internally |
| 15:29.52 | hexanol | that said, I believe the ringinuse parameter has no influence on this problem |
| 15:30.33 | Greenlight | It's been a while since I went over it, but I think it just effects the frequency of the problem |
| 15:30.34 | hexanol | ok, thank you for the info |
| 15:31.01 | Greenlight | No problems |
| 15:31.33 | hexanol | in my case, this is a bit delicate, since I'm also using some phones which doesn't behave correctly when two INVITE are received at the same time |
| 15:31.47 | hexanol | another question, any ideas on how to worked around it ? |
| 15:31.56 | jmetro | hand code queues |
| 15:32.03 | Greenlight | In my case I did workarounds in the AMI side |
| 15:32.18 | Greenlight | Yes, rather extreme, but write you're own queue |
| 15:32.31 | Greenlight | You may also have success with calllimit and GROUP_COUNT |
| 15:32.36 | hexanol | indeed, but I'm first looking for a quick fix, which is not exactly that |
| 15:32.53 | jmetro | the included queue system with asterisk has given me MANY headaches just with how it handles agent availablility |
| 15:32.58 | Greenlight | If the phones are *only* needing calls from queues, then calllimit may work |
| 15:33.18 | hexanol | ok |
| 15:33.29 | Greenlight | (I think that's the name of the option) |
| 15:33.42 | hexanol | yeah I know what you are talking about |
| 15:34.18 | Greenlight | I'm always suprised that this bug doesn't effect more people to be honest |
| 15:34.29 | jmetro | most probably only have 1 queue |
| 15:34.37 | Greenlight | It still happens |
| 15:34.50 | Greenlight | As in hexonals case |
| 15:34.58 | jmetro | hexonal had 2 queues i thought? |
| 15:35.11 | hexanol | yes, I have 2 queues |
| 15:35.17 | Greenlight | Oh |
| 15:35.45 | jmetro | imo it sounds like app_queue was coded poorly, and should set the phone to be in use much earlier |
| 15:35.56 | Greenlight | agrees |
| 15:36.13 | jmetro | theres a lot of things with the way app_queue handles phone availability that make me rage |
| 15:36.14 | Greenlight | My guess is they didn't want to lock *too* much |
| 15:36.15 | hexanol | when calls enter the same queue, the problem doesn't show up |
| 15:36.22 | Rahoul | thanks for your help bye |
| 15:37.05 | hexanol | yes, in fact, app_queue doesn't seem to share member between queue, so they are seen as independant |
| 15:37.23 | hexanol | and the member comes from devicestate callback |
| 15:37.40 | hexanol | the member *status |
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| 15:42.21 | Greenlight | I'm sure I've had it happen to calls in the *same* queue, although the timing had to be more precise |
| 15:48.06 | igcewieling | queues are proof that evil exists in the world. |
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| 15:50.52 | jmetro | Queue's are wonderfull in theory. I just wish there was an easier way to define your own strategy..like drag-dropping a flow chart. |
| 15:51.01 | jmetro | and then have it work * |
| 15:51.48 | igcewieling | Often a hunt group is a good solution for simple queue-like functionality. The problem is many people need complicated queue features. 8-| |
| 15:51.48 | jmetro | i dont know why they would deprecate strategies without a replacement either |
| 15:52.16 | Greenlight | I guess it's difficult to suit everyone |
| 15:52.28 | jmetro | i mean the simple "secretary first" strategy isnt implemented without wrangling |
| 15:52.39 | jmetro | or "top down" |
| 15:53.36 | Greenlight | One thing that I always get pointed out when I've moved someone to Asterisk from say Ayava or Cisco, is when a queue has a caller queued, and your on the phone. As soon as you hangup, your phone doesn't start to ring immediately again as with other PBXs |
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| 15:54.02 | igcewieling | Hmm? That is what hunt groups are best suited for. i.e. always calling a set of destinations always in the same order. |
| 15:54.31 | jmetro | yes, but they dont show up as "in a queue" and you cant have this "pool of queued callers" in a hunt group |
| 15:55.38 | igcewieling | You can code something similar in a dialplan or use a queue with one member (the lead TN in the hunt group) |
| 15:56.26 | igcewieling | the main thing you don't get with hunt groups is the sorts of queue stats some people want. |
| 15:56.35 | jmetro | that too |
| 15:57.01 | jmetro | its like app_queue needs to be rewritten every time we want something that isnt rrmemory |
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| 16:51.13 | newzen | Hello, there is any way to rotate outgoing lines in random way?? |
| 16:51.41 | newzen | means fxo line's |
| 16:51.56 | [TK]D-Fender | newzen: "core show function RAND" <- |
| 16:52.57 | jmetro | <--^---- |
| 16:56.44 | newzen | thanks |
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| 17:36.30 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 17:48.26 | ChannelZ | Want to wear out all the lines equally? |
| 17:49.50 | igcewieling | ChannelZ: likely wants to make sure the same line is not used all the time when one of the lines is out of service. |
| 17:50.11 | igcewieling | round robin with memory (R/r) is a better way to do that |
| 18:01.19 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 18:07.46 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
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| 18:38.54 | *** join/#asterisk Free99 (~Free99@ool-4350dd5c.dyn.optonline.net) |
| 18:40.25 | Free99 | if I wanted to dump sip debug to a file instead of the CLI, how would I do that? |
| 18:40.50 | navaismo | asterisk -rnvvvvvvdd | tee filename.txt |
| 18:41.18 | Free99 | ok, thanks navaismo. |
| 18:41.42 | *** join/#asterisk nightrid3r (~michel@62.205.85.88) |
| 18:44.10 | jmetro | Tee? |
| 18:44.32 | jmetro | oh. never saw that one before. Nice |
| 18:44.42 | Free99 | I'm having an issue where I keep trying to make calls with SIP phones (a linksys and Linphone) and I keep getting problems related to "Forbidden" when asterisk wants to pass the call to the client |
| 18:44.43 | Free99 | http://pastebin.com/Auwe2MGr |
| 18:45.01 | Free99 | I have nat on and no for directmedia.. |
| 18:47.26 | Free99 | jmetro: tee works, but it grabs everything, even control characters meant for the terminal window.. ymmv |
| 18:47.34 | Free99 | it worked tho |
| 18:52.05 | navaismo | -n suppossed to remove the color characters |
| 18:52.46 | Free99 | ah, forgot the n |
| 18:55.25 | navaismo | and your sip debug is incomplete |
| 18:57.05 | navaismo | hmmm -n isn't removing the colors |
| 18:57.38 | Free99 | thank goodness for pastebinit, found it in aptitude.. |
| 18:58.36 | Free99 | only problem is, it uses ubuntu's pastebin: http://paste.ubuntu.com/5605713/ |
| 18:58.43 | Free99 | sorry navaismo |
| 18:59.05 | navaismo | you can use this to remove color characters---> sed -i "s:\x1B\[[0-9;]*[mK]::g" filename.txt |
| 18:59.49 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 18:59.53 | Free99 | oh snap. Just realized this softphone is trying to register as 192.168.1.169... that's what is happening, right? |
| 19:05.59 | navaismo | I see one call making progress, but then they responf with: SIP/2.0 403 Forbidden - Bad Account/PIN |
| 19:06.34 | navaismo | so you may check that "ACCOUNT/PIN" thing |
| 19:06.37 | navaismo | or funds |
| 19:07.03 | Free99 | hmm. that's supposed to be my termination provider... they let me register though. Ah, I guess that makes sense |
| 19:07.30 | Free99 | how would I force a re-register? |
| 19:07.36 | Free99 | on my pbx I mean |
| 19:07.47 | igcewieling | a sip reload sill make it reregister |
| 19:08.16 | Free99 | "sip reload sill"? |
| 19:08.27 | Free99 | why sill? |
| 19:08.33 | navaismo | only sip reload |
| 19:12.40 | Free99 | oh, derp. thanks navaismo |
| 19:12.46 | *** join/#asterisk stanreg (2679fd0b@gateway/web/freenode/ip.38.121.253.11) |
| 19:13.36 | stanreg | Hey. What would you guys recommend as a low-cost upgrade to an aging Linksys PAP2T? Preferably something that does hole punching on its own.. |
| 19:14.06 | igcewieling | sip reload will make it register. I can't type today |
| 19:15.41 | *** join/#asterisk vlad_starkov (~vlad_star@188.123.248.55) |
| 19:20.02 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:20.39 | Free99 | I just don't get why asterisk is saying that the caller is forbidden |
| 19:21.09 | Free99 | Received response: "Forbidden" from '"4441012002" <sip:4441012002@176.58.108.118>;tag=as4676745a' |
| 19:21.22 | Free99 | though the caller is actually at a different IP |
| 19:21.31 | igcewieling | that is almost always because your peer is not matching |
| 19:21.53 | igcewieling | sip debug should show what peer is matching |
| 19:21.55 | Free99 | igcewieling, what do you mean by "not matching"? |
| 19:22.39 | igcewieling | I mean the inbound call is not matching an entry in sip.conf so the call is rejected. |
| 19:23.28 | igcewieling | is this inbound or outbound calling (from the perspective of Asterisk) |
| 19:24.29 | Free99 | outbound I guess. |
| 19:24.37 | Free99 | http://paste.ubuntu.com/5605806/ |
| 19:24.52 | Free99 | I get sound back from my * box, but the call just fails |
| 19:25.20 | igcewieling | Oh, a2billing. You are on your own. |
| 19:26.12 | navaismo | haha |
| 19:26.14 | Free99 | hmm.. you think a2billing is responsible for the "Forbidden" message? |
| 19:26.17 | igcewieling | one key to solving Asterisk problems is to make everything simple so the issue can be diagnosed. You can't do that with some huge billing system in the mix. I wish you the best of luck. Putting you on /ignore so I don't accidentally answer something |
| 19:26.42 | Free99 | wow thanks igcewieling. |
| 19:29.42 | Free99 | I'll just try figuring that issue out on my own I suppose |
| 19:29.57 | jmetro | i would check into the whole peer matching thing |
| 19:29.58 | navaismo | I stll guessing that your forbidden is part of your "account/pin" just in the cli you dont see extra sip message |
| 19:30.42 | Free99 | hmm. I noticed that I have two clients at the same IP address trying to use port 5060 |
| 19:30.58 | Free99 | perhaps that is related? |
| 19:33.54 | Free99 | let's skip that for right now, I have a problem I know only pertains to asterisk itself. I have tried linphone and qutecom to talk to my server, I have a part on the dialplan where extension 123 is supposed to playback tt-weasels |
| 19:34.17 | Free99 | if I dial with the linksys units, I hear it fine. If I play with the softphones however, nothing |
| 19:34.40 | *** join/#asterisk mathi (Matt@92.249.121.92) |
| 19:34.41 | mathi | hi |
| 19:35.29 | mathi | is it possible to have a "+" character in Dial() for international call ? e.g. Dial(SIP/+32494873465@VoIPProviderOutbound,15) |
| 19:35.34 | mathi | would that be valid ? |
| 19:38.54 | Free99 | http://paste.ubuntu.com/5605856/ <- plays back tt-weasels, no audio on the client |
| 19:42.19 | leifmadsen | mathi: yes, that is valid |
| 19:46.03 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:55.17 | Free99 | is linphone a good sip client for windows? |
| 19:58.18 | jmetro | i like 3cx phone personally |
| 20:01.35 | *** part/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 20:01.40 | Free99 | jmetro, is it open-source? |
| 20:01.46 | Free99 | or, lemme go look lol |
| 20:02.05 | jmetro | its not open source im pretty sure |
| 20:02.07 | jmetro | 3cx is commercial |
| 20:02.11 | jmetro | but they have a free softphone |
| 20:04.48 | *** join/#asterisk stanreg_ (2679fd0b@gateway/web/freenode/ip.38.121.253.11) |
| 20:05.54 | stanreg_ | What would you guys recommend as a low-cost upgrade to an aging Linksys PAP2T ATA? Preferably something that easily punches through NAT.. |
| 20:06.58 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 20:07.25 | igcewieling | Asterisk includes over 1500 sound files and there is never one I need. 8-| |
| 20:07.46 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:07.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:07.46 | Free99 | tt-weasels solves all problems lol |
| 20:08.28 | jmetro | stanreg something wrong with the cisco pap2t which is just the linksys pap2t? it works well. |
| 20:08.31 | Free99 | weird, linphone works on linux, but not on windows 7 |
| 20:08.46 | jmetro | i like extension-trick |
| 20:08.52 | Free99 | that is, I actually hear my * box |
| 20:09.10 | igcewieling | I wanted something like "Your local carrier is" and "Your toll carrier is" |
| 20:09.21 | Free99 | oh yeah, jmetro you were the one who showed me (S0:<phone number>) right? |
| 20:09.33 | igcewieling | We have a recording of allison saying our company name 8-) |
| 20:09.58 | jmetro | Free99: whats that? |
| 20:09.58 | stanreg_ | jmetro: I've noticed that having more than one pap2t on the same network tends to create disconnects; haven't you? |
| 20:10.28 | jmetro | stanreg_: negative, we have at least 10-20 of them on our network. Maybe on different subnets helps. |
| 20:11.01 | stanreg_ | jmetro: hmm, different subnets would additional equipment, no? |
| 20:11.06 | stanreg_ | would imply* |
| 20:11.12 | jmetro | negative, one armed routing. |
| 20:12.30 | jmetro | one router services multiple subnets out of one interface, switches handle multiple subnets on one physical switch using vlans |
| 20:12.47 | stanreg_ | sounds overly complex for a home. |
| 20:13.00 | jmetro | for home, i doubt you have the right equipment yeah. |
| 20:14.17 | *** join/#asterisk kuruption (kuruption@laffs.at.the.lol.lawlz.lulz.liberlawls.com) |
| 20:15.28 | igcewieling | there is no reason having multiple PAPs on a network would be a problem. If your router is brain dead or has SIP ALG or SPI enabled or you do something stupid like port forwarding then having more than one of ANY SIP device could cause random disconnects |
| 20:15.41 | navaismo | Free99, lastest version of linphone cant work try downgrading only in windows |
| 20:15.58 | jmetro | Free99: whats wrong with 3cx free softphone? just wondering. |
| 20:16.01 | jmetro | noone seems to like it |
| 20:16.02 | Free99 | oh man, for real? where did you find that? |
| 20:16.08 | jmetro | but i had monumental success with it |
| 20:16.27 | Free99 | jmetro: I want to tune the UI a little, make chatting and stuff easier |
| 20:16.30 | stanreg_ | See, my pap2t's state often switches to "Can't connect to login server", until I soft-reset by going to: [ip]/admin/reboot -- I'm trying to avoid this step. Not everyone at home is tech-savvy :\ |
| 20:16.37 | Free99 | can't do that with closed source you know? |
| 20:17.20 | jmetro | Free99: you could also script that for them |
| 20:17.23 | jmetro | wrong person |
| 20:17.27 | jmetro | stanreg_: you could script that. |
| 20:17.29 | igcewieling | stanreg_: do you do any port forwarding on your router? |
| 20:17.36 | stanreg_ | igcewieling: nothing fancy here; good ol' openwrt. negative. |
| 20:17.51 | jmetro | Free99: 3cx phone actually supports themes from what i remember |
| 20:17.57 | jmetro | for UI customization |
| 20:18.29 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:18.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:18.52 | Free99 | other thing is, unified codebase... but I'll check it out actually |
| 20:19.11 | igcewieling | stanreg_: you have the latest firmware? |
| 20:20.27 | stanreg_ | igcewieling: yep |
| 20:20.44 | stanreg_ | jmetro: wouldn't that imply having a computer on at all times? |
| 20:20.45 | navaismo | 3cx--->windows we preffer opnsource stuff hahaha |
| 20:22.04 | Free99 | I mean I'm a linux sysmin, so kinda yeah lol |
| 20:22.19 | Free99 | but we have windows clients and stuff, so.. |
| 20:23.14 | jmetro | stanreg_: I meant make a little script for your family to run on their PC's when the issue happens. i guess you could stick it on the asterisk box too. |
| 20:23.25 | jmetro | since that is inherently running all the time |
| 20:24.41 | stanreg_ | jmetro: Hmm.. Ideally, the pap2t would "just work" ;) |
| 20:25.04 | stanreg_ | ..but what is this, surely I am asking too much! :D |
| 20:25.09 | Free99 | stanreg_: I've only started using pap2t recently, they work really well |
| 20:25.16 | Free99 | better than most softphones thus far |
| 20:25.27 | *** join/#asterisk FonJockey (~FonJockey@173-132-41-133.pools.spcsdns.net) |
| 20:26.18 | stanreg_ | Free99: I don't doubt it ... though they're almost a decade old now and, as you can tell by my experience, not always reliable. |
| 20:26.49 | Free99 | was about to ask, "what's the alternative", when I remembered that's the reason you visited lol |
| 20:27.29 | jmetro | the Spa115 i think it is? is not bad at all |
| 20:28.50 | jmetro | spa112 |
| 20:29.07 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:29.07 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:29.15 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 20:33.34 | stanreg_ | spa112 looks interesting indeed |
| 20:43.37 | mathi | hey |
| 20:43.45 | mathi | what do you advice as outbound sip provider ? |
| 20:44.05 | mathi | leifmadsen, thanks for your response by the way)) |
| 20:47.37 | jmetro | stanreg_: the spa series is all rebranded linksys but its new and it works great. |
| 20:48.05 | stanreg_ | jmetro: I wonder how different it trully is on the inside (pap2t vs spa112) |
| 20:49.40 | jmetro | probably not very but it is working great on our network. i havent seen one fail and the biggest issue we have.... i cant even remember it. |
| 20:50.03 | jmetro | the Snom PA can also function as a Pap2t |
| 20:55.25 | igcewieling | mathi: vitelity |
| 21:07.55 | navaismo | flowroute and last voip.ms |
| 21:13.23 | *** join/#asterisk Dovid (~Dovid@48.sub-70-192-132.myvzw.com) |
| 21:17.06 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
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| 21:45.40 | *** join/#asterisk alhemicar (~alhemicar@178.77.23.11) |
| 21:50.44 | Kobaz | [2013-04-26 17:50:15] WARNING[17516]: db.c:329 ast_db_put: Couldn't execute statment 'INSERT OR REPLACE INTO astdb (key, value) VALUES (?, ?)': SQL logic error or missing database |
| 21:50.48 | Kobaz | [2013-04-26 17:50:15] WARNING[17509]: db.c:285 db_execute_sql: Error executing SQL: database is locked |
| 21:50.57 | Kobaz | sqlite is throwing errors |
| 21:51.39 | cusco | or replace? |
| 21:51.42 | cusco | ow, sqlite |
| 21:51.50 | cusco | not sure about syntax |
| 21:52.27 | cusco | database is locked |
| 21:52.30 | Kobaz | yeah |
| 21:52.35 | Kobaz | not sure how to unlock it |
| 21:52.46 | cusco | my sqlite knowlege is short on this matter |
| 21:53.11 | cusco | google shows something like |
| 21:53.18 | cusco | fuser / lsof |
| 21:53.22 | cusco | to check the pid using it |
| 21:53.24 | cusco | (asterisk) |
| 21:53.26 | cusco | and kill the pid |
| 21:53.46 | cusco | also "echo ".dump" | sqlite old.db | sqlite new.db" |
| 21:53.58 | cusco | http://www.sqlite.org/cvstrac/wiki?p=DatabaseIsLocked |
| 21:54.03 | *** join/#asterisk vlad_starkov (~vlad_star@77.232.28.64) |
| 21:54.25 | cusco | "But the SELECT is not really complete until sqlite3_reset() or sqlite3_finalize() have been called." |
| 21:54.51 | Free99 | hey guys. I've cut a2billing out of the dialplan but still can't make calls |
| 21:54.53 | Free99 | http://paste.ubuntu.com/5606253/ |
| 21:55.00 | cusco | "fcntl(3, F_SETLK call on DB file fails. This could be caused by an NFS locking issue, for example. One solution for this issue, is to mv the DB away, and copy it back so that it has a new Inode value." |
| 21:55.08 | Kobaz | well i restated asterisk |
| 21:55.11 | Kobaz | and the sqlite db isn't locked |
| 21:55.14 | Kobaz | i think it's a syntax error |
| 21:55.15 | cusco | :) |
| 21:55.23 | Kobaz | i think 'INSERT OR REPLACE INTO' is invalid |
| 21:55.24 | Free99 | oh shoot, forgot to turn on sip debug, hang on |
| 21:55.28 | cusco | asterisk failed to finish the ``transaction´´ |
| 21:56.30 | Free99 | http://paste.ubuntu.com/5606262/ |
| 21:56.36 | Kobaz | oooo |
| 21:56.51 | Kobaz | # sqlite -version |
| 21:56.52 | Kobaz | 2.8.17 |
| 21:56.57 | Kobaz | it is a syntax problem |
| 21:56.59 | cusco | oooooo! |
| 21:57.03 | Kobaz | asterisk assumes sqlite3 |
| 21:57.15 | cusco | ofcourse, dunno how configure script didn't check that |
| 21:57.16 | cusco | :P |
| 21:57.24 | cusco | its a big change from 2 to 3 |
| 22:00.10 | *** join/#asterisk jsjc (~Adium@112.Red-83-54-23.dynamicIP.rima-tde.net) |
| 22:00.31 | mathi | actually i found i still had a voipcheap account with some credits |
| 22:00.38 | mathi | but when I try to do a call |
| 22:00.41 | mathi | I have no audio |
| 22:00.51 | mathi | neither the caller nor me who receives the call |
| 22:01.05 | Kobaz | okay i apt-get installed sqlite3 |
| 22:01.06 | Kobaz | hmm |
| 22:01.11 | Kobaz | i can read the db file just f ine |
| 22:01.20 | Kobaz | asterisk still complaining |
| 22:01.41 | cusco | recreate the db? |
| 22:01.52 | Kobaz | the db is fine |
| 22:01.55 | Kobaz | sqlite3 can read/write it |
| 22:03.05 | Kobaz | i dunno maybe |
| 22:03.15 | Kobaz | i'll wipe it out instead of doing ast2db2sqlite3 |
| 22:03.16 | Free99 | Kobaz: I've made sqlite based print servers in the past, nothing but mysterious bugs. I dunno what your usage is, but disk-lag may be a contributing factor (as it was in my case) |
| 22:03.19 | *** join/#asterisk andrewyager (~andrewyag@CPE-144-132-193-27.nsw.bigpond.net.au) |
| 22:03.43 | Kobaz | hey well hah, that worked |
| 22:03.52 | Kobaz | oh asterisk auto converted it |
| 22:04.22 | Kobaz | okay yay |
| 22:04.53 | Kobaz | people need to stop working so i can restart asterisk |
| 22:05.02 | Kobaz | need to do two other servers |
| 22:05.48 | Free99 | is there a way to make asterisk place a call without needing /dev/dsp? |
| 22:05.55 | Free99 | from the cli I mean |
| 22:06.06 | Qwell | channel originate |
| 22:06.09 | Kobaz | yeah |
| 22:06.10 | Kobaz | originate |
| 22:06.11 | Kobaz | yeap |
| 22:07.52 | navaismo | Free99, have you checked your peer seetings(from provider ) and your funds and if you can dial that number--->SIP/2.0 403 Forbidden - Bad Account/PIN |
| 22:08.20 | Free99 | navaismo, double checked, I have $15 on the account too. |
| 22:08.25 | Free99 | no a2billing either |
| 22:09.02 | Qwell | Free99: Where's the SIP debug? |
| 22:09.08 | Free99 | http://paste.ubuntu.com/5606262/ |
| 22:09.41 | navaismo | Free99, your call is reaching the Net2Phone ippbx at but they answer to you with that --->SIP/2.0 403 Forbidden - Bad Account/PIN |
| 22:10.02 | navaismo | You can see the 'making progress pass' and then forbidden |
| 22:10.25 | Free99 | I'd like to stop annoying you guys with my sip debug logs, but I need to know how they work. How do you know what these various messages mean? |
| 22:10.25 | navaismo | your peer are sending the fromuser/fromdomain or whatever |
| 22:10.38 | Qwell | Free99: Ask your provider. They're sending it. |
| 22:11.09 | Free99 | not just the 403 errors, but the sip messages in general |
| 22:11.27 | Free99 | is there a lookup somewhere for these, or just enough time looking at 'em? |
| 22:11.56 | navaismo | http://www.ietf.org/rfc/rfc3261.txt |
| 22:13.47 | Qwell | Nobody here can tell you what it means. We don't have access to their systems. Ask them. |
| 22:16.23 | Free99 | qwell: I meant the actual sip dialog and such hehe. I'm going to get on the horn with them. Thanks for the help |
| 22:18.51 | Qwell | df |
| 22:18.54 | Qwell | erm, yeah |
| 22:19.12 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 22:20.10 | *** join/#asterisk sjs205 (~sjs205@host81-155-250-128.range81-155.btcentralplus.com) |
| 22:29.54 | Kobaz | df du bc |
| 22:31.13 | Kobaz | come oooooon |
| 22:31.19 | Kobaz | get offfff the pphooooooooone |
| 22:31.41 | Kobaz | oh cool |
| 22:36.22 | Kobaz | hey hmm |
| 22:36.36 | Kobaz | in 11 is there some new option that controls whether or not to resend mwi on REGISTER ? |
| 22:41.10 | *** join/#asterisk vlad_starkov (~vlad_star@77.232.28.64) |
| 22:50.01 | Kobaz | ugh, this new mwi-on-register in 11 is really annoying |
| 22:54.13 | *** part/#asterisk alhemicar (~alhemicar@178.77.23.11) |
| 22:55.53 | ChannelZ | well you can tell it to only do it if the phone subscribes specifically.. |
| 22:58.01 | Free99 | what is MWI? Message waiting indicator? |
| 22:58.37 | cusco | voicemail stuff |
| 22:59.23 | Free99 | slow DNS stinks :-/ |
| 23:00.29 | Kobaz | yeah |
| 23:02.04 | cusco | its supposed to be fast |
| 23:04.26 | *** join/#asterisk Quest (~syncsys@139.190.90.234) |
| 23:04.27 | Quest | what POE switch is best for ip phones cisco 7960 ? |
| 23:05.04 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 23:05.58 | ChannelZ | Anything with enough actual power to run the number of handsets you have |
| 23:06.36 | Kobaz | yeah |
| 23:06.41 | ChannelZ | I have a Netgear GS748 I'm pretty pleased with |
| 23:07.16 | ChannelZ | GS748TP I guess I should say |
| 23:08.30 | Quest | need cisco one |
| 23:08.48 | cusco | eeq |
| 23:08.52 | cusco | routerboard.com |
| 23:08.53 | cusco | :p |
| 23:09.14 | ChannelZ | eh? |
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| 23:24.37 | stanreg_ | jmetro: wondering out loud -- do you think enabling uPnP on the router might allow multiple PAP2T to work properly on the same network? |
| 23:37.56 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 23:48.03 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 23:55.29 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 23:56.24 | *** join/#asterisk FonJockey (~FonJockey@173-133-156-123.pools.spcsdns.net) |
| 23:58.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 23:58.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
