| 00:03.57 | *** join/#asterisk sustav (~vpp@76.73.166.16) |
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| 00:50.36 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 00:51.16 | pzn | Hi. I need to modify incoming callerIDs, can I configure asterisk to execute a perl script with the callerID? |
| 01:03.51 | *** join/#asterisk deo (~dnepangue@58.71.19.178) |
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| 01:09.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:18.17 | jpsharp | pzn: You can execte an AGI, but what do you need to do with the caller ID. You might be able to do it with the dialplan. |
| 01:19.15 | pzn | jpsharp, I need to search a postgresql database to replace callerid when needed |
| 01:19.22 | pzn | jpsharp, I'll google for AGI and do it |
| 01:23.18 | pzn | jpsharp, nice! AGI documentation is exactly what I need! |
| 01:24.21 | jpsharp | Excellent. Or you could use ODBC to talk directly to Postgres, but that requires Asterisk to be built with ODBC support. |
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| 02:00.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 02:28.14 | webbeaver | hi, which hardware do i need for starting with asterisk? |
| 02:30.34 | jpsharp | A PC. |
| 02:30.44 | jpsharp | One that can run Linux. |
| 02:33.18 | webbeaver | thanks, and do i need a voip provider? |
| 02:33.45 | jpsharp | Only if you want to talk to the outside world. |
| 02:34.05 | webbeaver | ok thank you |
| 02:34.12 | webbeaver | have fun |
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| 03:43.24 | volga629 | what can cause for ignoring 'video' media offer because port number is zero |
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| 04:08.08 | ChannelZ | "A port number of zero in the offer indicates that the stream is offered but MUST NOT be used." |
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| 04:10.43 | ChannelZ | Perhaps your device is misconfigured? |
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| 06:18.22 | schmidts | good morning |
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| 06:23.33 | v0lZy | lo |
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| 07:11.54 | gavimobile | folks, this has stoped working for me this morning. [motekPCtimeconditions] |
| 07:11.55 | gavimobile | exten => start,1,Answer() |
| 07:11.55 | gavimobile | <PROTECTED> |
| 07:11.55 | gavimobile | <PROTECTED> |
| 07:13.00 | gavimobile | my incoming did Goto this context |
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| 07:26.37 | v0lZy | and then? |
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| 07:40.51 | ChannelZ | gavimobile: what did it do? |
| 07:44.56 | gavimobile | ChannelZ: wow, after playing with 100 different scenarios of the above code, taking away Answer() seemed to solve the problem |
| 07:45.35 | gavimobile | very strange that just this morning it deciced not to work.. its been working fine for the last week or so since I made my server live |
| 07:46.18 | gavimobile | wtf.. its now not working |
| 07:46.55 | gavimobile | ChannelZ: what happenes is I hear dead air |
| 07:47.33 | ChannelZ | You have some other issue |
| 07:48.01 | ChannelZ | Completely unrelated to the piece of dialplan you posted |
| 07:48.52 | gavimobile | ChannelZ: well that narrows it down |
| 07:49.31 | gavimobile | ChannelZ: how can I narrow down for you the sip debug output |
| 07:49.39 | gavimobile | putty cannot log all of it |
| 07:50.16 | bulkorok | asterisk -rvvvvv | tee /tmp/astcli.txt |
| 07:50.18 | gavimobile | how about sip set debug peer my-incoming-trunk-name |
| 07:50.28 | gavimobile | bulkorok: im going to save that line of code |
| 07:50.30 | ChannelZ | It depends what is actually failing. If you are having audio problems it's probably firewall issues |
| 07:51.07 | gavimobile | well in my dialplan when I call 300 it goes to my queue |
| 07:51.29 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 07:51.32 | gavimobile | dialing 300 works, however there seems to be a problem between the incomming call and the queue |
| 07:51.32 | ChannelZ | You should define your problems a little better when you ask however; The first question you asked seemed to imply GotoIfTime wasn't doing what you expected, yet come to find out your actual problem seems to be silence |
| 07:51.52 | ChannelZ | And now we're hearing about queues |
| 07:52.45 | gavimobile | ChannelZ: yes, agreed, well from my dialplan section from the begening of the incoming call it goes to a time condition. so i commented the time condition and put a playback() there which seems to work flawless |
| 07:52.46 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 07:53.09 | gavimobile | I did this all the way to the time condition |
| 07:54.40 | ChannelZ | So it always works if you have a Playback somewhere early in the call? |
| 07:55.25 | gavimobile | I am pretty sure, give me just a moment to double check |
| 07:55.43 | ChannelZ | smells of misconfigured NAT. Is Asterisk behind a firewall? |
| 08:00.56 | gavimobile | ChannelZ: yes, from my router.. playback is intermittint also :-( |
| 08:03.00 | fling | ChannelZ: hello! :p what dir do I need to place my sounds? |
| 08:03.10 | fling | ChannelZ: I have recorded a digital girl |
| 08:03.39 | gavimobile | fling: /var/lib/asterisk/sounds/custom/ is a good place |
| 08:03.47 | fling | ChannelZ: she says "If you are female press 1, if you are male press 2" |
| 08:03.54 | fling | gavimobile: ok, good, thanks |
| 08:04.11 | gavimobile | fling: Playback(custom/mysound) |
| 08:04.22 | fling | sweet |
| 08:04.35 | gavimobile | fling: don't forget to give the proper permission to your files |
| 08:04.52 | fling | chown asterisk:asterisk ? |
| 08:05.32 | gavimobile | could someone in the channel answer that for us. im not one to speak when it comes to permissions |
| 08:05.32 | fling | and chmod 770 custom … |
| 08:05.37 | bulkorok | if asterisk is running under this user it should work |
| 08:05.48 | gavimobile | thanks bulkorok |
| 08:06.20 | gavimobile | ChannelZ: I could use your help here |
| 08:07.26 | fling | gavimobile: use Dial |
| 08:07.29 | ChannelZ | You must properly set externaddr and localnet in sip.conf. You must port-forward the UDP ports in the range listed in rtp.conf to your Asterisk server. And pretty much all UDP out from Asterisk must be allowed. |
| 08:07.58 | fling | ChannelZ: hey! my asterisk works fine now, thanks for you previous help :p |
| 08:08.26 | ChannelZ | I don't remember what your issue was but if I helped, you're welcome |
| 08:08.35 | gavimobile | ChannelZ: they are properly set in sip.conf. my ports are forwarded as listed in rtp (however tcp&& udp are being forwarded on theose ports) |
| 08:08.58 | fling | ChannelZ: something about dial, moh, etc |
| 08:09.34 | gavimobile | externaddress=mynetwork's external ip address which is static |
| 08:09.57 | gavimobile | and localnet is 192.168.0.0 if my default gateway is 192.168.0.1 |
| 08:11.12 | ChannelZ | well it's "externaddr" and localnet would be 192.168.0.0/16 to be exact |
| 08:11.13 | gavimobile | regarding the tcp && udp.. I've never had trouble with forwarding both |
| 08:11.43 | gavimobile | so I should add a /16 at the end of the line in localnet? |
| 08:12.07 | gavimobile | my bad, the value actually looks like this right now localnet=192.168.0.0/255.255.255.0 |
| 08:12.09 | ChannelZ | ideally |
| 08:12.26 | gavimobile | ok.. ill cahnge it now |
| 08:12.42 | *** join/#asterisk sbszulu (~Sibusiso_@41.48.192.112) |
| 08:12.55 | gavimobile | done |
| 08:13.00 | fling | gavimobile: this works for me > localnet=10.0.0.0/8 |
| 08:13.04 | ChannelZ | If thinks work reliably having asterisk Playback something early on, then do that. (What does your dialplan normally do?) |
| 08:13.22 | ChannelZ | fling: except that's a different network than his seems to be |
| 08:13.46 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 08:13.50 | fling | ChannelZ: right, but looks like it is not needed to specify 255x long mask, /24 should be enough |
| 08:13.59 | gavimobile | ChannelZ: apparently things aren't working "reliably" earlier in my dialplan ..my assumption as wrong as I mentioned above |
| 08:14.12 | fling | localnet=192.168.0.0/24 ? |
| 08:14.15 | ChannelZ | Anyway I've seen Asterisk do some wierd things when NAT is involved, not sending RTP to the right place in certain circumstances unless you prime it with some audio |
| 08:14.55 | fling | ChannelZ: I have some strange things too if localnet specified |
| 08:14.58 | ChannelZ | fling: that'd work too assuming everything was only on 192.168.0 |
| 08:15.02 | gavimobile | I seem to have an intermittant problem and I have a weird feeling that possibly channels are still being used and not hung up. but im no pro |
| 08:17.22 | gavimobile | I duplicated the issue after the changes ChannelZ |
| 08:18.30 | fling | ffmpeg -i infile.flac -f alaw outfile.alaw |
| 08:18.37 | ChannelZ | try doing rtp set debug on and calling in again, see if it seems to be receiving RTP from the correct IP and sending it back out to the correct IP as well |
| 08:18.46 | fling | ^ am I converting right? |
| 08:20.27 | gavimobile | ChannelZ: k, ill try that no, in the meantime here is my issue in words.. note the third call wasn't logged but it could be the call is in progress http://pastebin.com/3crCh0jc |
| 08:21.49 | gavimobile | ChannelZ: my output is as follows Got RTP packet from 82.166.67.33:8212 (type 00, seq 012961, ts 1775812275, len 000160) |
| 08:21.49 | gavimobile | Sent RTP packet to 82.166.67.33:8212 (type 00, seq 055490, ts 072640, len 000160) |
| 08:22.14 | gavimobile | the ip provided to my from my did provider is 82.166.66.43 |
| 08:22.23 | gavimobile | and 82.166.67.43 |
| 08:22.36 | gavimobile | does this say anything |
| 08:25.16 | fling | looks like this is right > for file in * ; do ffmpeg -i $file -acodec pcm_alaw -ar 8000 -f alaw ${file/flac/alaw} ; done |
| 08:27.32 | *** join/#asterisk Anderson-D (~anderson@mail.primocollect.com.ua) |
| 08:27.33 | Anderson-D | Hello |
| 08:28.10 | Anderson-D | Asterisk at max verbosity throws me a bunch of "Skipping field 'xxxx'='yyy'" during update of sip_users. Any ideas? |
| 08:28.47 | ChannelZ | Is something not working? |
| 08:29.41 | gavimobile | ChannelZ: you asking me? |
| 08:29.52 | ChannelZ | gavimobile: Dunno, I assume they have multiple IPs and that one is what is listed in their SDP when the call gets setup. It's good news that it's sending/receiving packets anyway. But if you're saying during that call you had no audio or one-way audio then something else is screwed up, maybe your device. |
| 08:30.09 | ChannelZ | But I also thought you were calling IN, not out as your paste showed. |
| 08:30.15 | *** join/#asterisk Corydon76-home (two@c-69-137-80-31.hsd1.tn.comcast.net) |
| 08:30.15 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 08:30.37 | gavimobile | ChannelZ: which device? my current device away from the office is a softphone |
| 08:30.52 | ChannelZ | yeah |
| 08:31.10 | Anderson-D | Also, what does 'key field `lastms` doesn't exist. update will fail' mean? |
| 08:31.14 | Anderson-D | That field IS there |
| 08:31.55 | gavimobile | ChannelZ: well playback should be showwing in the cli that its playing regardless of my device |
| 08:32.05 | ChannelZ | gavimobile: now I'm even more confused.. you're dialing with a softphone that's not local to you? |
| 08:32.37 | gavimobile | im dialing to myself yes, im dialing not locally though |
| 08:33.03 | ChannelZ | ....huh? |
| 08:33.06 | gavimobile | my queue is say 300, im dialing my full telephone number |
| 08:33.49 | ChannelZ | I'm talking physically local. |
| 08:33.59 | ChannelZ | Where is the phone you're testing with? Where is Asterisk in relation to it? |
| 08:34.58 | gavimobile | ChannelZ: queue should be playing moh when a inbound call comes in.. is that the asterisk relation? |
| 08:35.21 | ChannelZ | my head hurts. |
| 08:36.17 | *** join/#asterisk timahvo1 (~rogue@41.215.82.210) |
| 08:36.58 | gavimobile | im using a softphone app from apple's appstore for testing right now |
| 08:37.50 | gavimobile | ill try a different softphone and im sure ill be able to duplicate the issue, its not an issue with my softphone |
| 08:38.54 | gavimobile | yup.. still causing trouble |
| 08:39.54 | gavimobile | can we go back a few steps to allow me to try and better explain myself to you? |
| 08:40.20 | gavimobile | im supprised you didn't ask for a sip debug yet |
| 08:40.34 | ChannelZ | Is the softphone on a LAN? Is that LAN the same LAN your Asterisk server is on? |
| 08:41.05 | gavimobile | ChannelZ: no |
| 08:41.16 | gavimobile | I am currently out of my office |
| 08:41.43 | ChannelZ | So you are testing on a softphone that is remote to Asterisk. And I assume you're behind a NAT/firewall too. |
| 08:42.45 | gavimobile | that correct to both of your question |
| 08:43.40 | ChannelZ | so then you're using that to call out through your ITSP back to yourself? |
| 08:45.30 | gavimobile | ChannelZ: almost correct |
| 08:46.16 | gavimobile | I am using that (my remote softphones which are not in my asterisks lan) to call through one itsp of mine to another which is also mine |
| 08:46.24 | gavimobile | hope I didn't confuse you there |
| 08:46.31 | gavimobile | im using trunk 2 to call trunk1 |
| 08:46.42 | gavimobile | I only use trunk1 for incomming |
| 08:47.34 | ChannelZ | ok so you're trying to debug an audio problem that could be on any one of 3 separate call legs. I stand no chance here |
| 08:48.48 | gavimobile | call legs? |
| 08:49.05 | gavimobile | why not |
| 08:51.07 | gavimobile | there is no audio problem |
| 08:51.31 | gavimobile | I don't explain unless you ask me to. than I explain to the best of my ability |
| 08:51.58 | gavimobile | I don't have audio problems |
| 08:55.14 | ChannelZ | <gavimobile> ChannelZ: what happenes is I hear dead air |
| 08:55.19 | gavimobile | ChannelZ: check this out |
| 08:55.25 | gavimobile | ChannelZ: http://pastebin.com/TUy8wBSd |
| 08:55.29 | gavimobile | before you click |
| 08:55.42 | gavimobile | I did 4 consecutive cals with a different remote softphone right now |
| 08:55.51 | gavimobile | all the calls go through |
| 08:56.36 | *** join/#asterisk niluje (~niluje@82.227.67.242) |
| 08:56.36 | gavimobile | the 4th call though went through as well, however I didn't hear the playback as the first 3 calls had.. if I answered the call there is no audio problem |
| 08:57.06 | gavimobile | yes, dead air... until I answer |
| 08:57.23 | bulkorok | doesn't a call have to be answered before playback!? |
| 08:58.28 | bulkorok | playback is RTP and if the call is not answered (SIP 200 OK) the rtp-session won't be established.... so playback doesn't know where to send the packets... |
| 08:58.38 | gavimobile | bulkorok: regardless, why did it work the first 3 times |
| 08:58.59 | bulkorok | with different softphones? |
| 08:59.06 | bulkorok | could be early media... |
| 08:59.24 | gavimobile | doesn't matter which softphone but in these 4 examples they were all the same softphone (zoiper) |
| 08:59.26 | ChannelZ | I don't even see your queue. I have no idea what is or isn't working. |
| 08:59.56 | *** join/#asterisk wonderworld (~ww@dsdf-4db5c653.pool.mediaWays.net) |
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| 09:00.12 | gavimobile | why didn't my custom file get played in the 4th call and why doesn't it show that the call has been received or forwarded in the 4th call like it does in the first 3 examples? |
| 09:00.17 | ChannelZ | anyway it's way past my bedtime |
| 09:00.50 | gavimobile | the only thing it shows in the 4th example is Locally bridging SIP/0000FFFF0000-000000a0 and SIP/didlogic-trunk-000000a1 |
| 09:00.58 | gavimobile | ChannelZ: thanks for your time |
| 09:01.27 | *** join/#asterisk Corydon76-home (red@c-69-137-80-31.hsd1.tn.comcast.net) |
| 09:01.27 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 09:02.12 | gavimobile | it shows it in the other 3 examples also, but after that it stops to show information... but the call is still in progress and I know this because the queue forwards the call to my cellphone number (gsm). and I get the call |
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| 09:47.28 | zamba | what is basically the difference between To: <number> and To: <number>@<hostnameofasterisk>? |
| 09:47.33 | zamba | in a sip invite? |
| 09:49.59 | bulkorok | http://www.packetizer.com/rfc/rfc3261/ point 8.1.1.2 |
| 09:54.08 | zamba | because i have a problem with my polycom phone is that it doesn't include the hostname when dialing from its list of missing calls |
| 09:54.26 | zamba | and this list is the full number, as in +4732342333 |
| 09:54.53 | zamba | i see this works on my softphone and when comparing the invite i see that my softphone dials with the hostname as well |
| 09:56.12 | zamba | "Looking for s in standard_sip (domain +4741921679)" |
| 09:56.18 | zamba | i guess this is the problem |
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| 12:19.21 | pietro | hello |
| 12:19.51 | pietro | why nobody take care this issue https://issues.asterisk.org/jira/browse/ASTERISK-18345 ? |
| 12:23.48 | *** join/#asterisk domi (~Thunderbi@mail.tas.de) |
| 12:24.52 | domi | is there a way to check if a sip-peer exist, without getting a warning when not, like ChanIsAvail does? |
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| 12:40.40 | leifmadsen | domi: look at the SIPPEER() dialplan function |
| 12:40.44 | leifmadsen | or CHANNEL() |
| 12:41.34 | leifmadsen | pietro: guess it hasn't gotten noticed by someone with commit access. Is there a reviewboard issue for the patch? That'll need to happen first. |
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| 12:42.10 | leifmadsen | malcolmd: can you review ASTERISK-18345 and make sure it has the correct priority? |
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| 12:42.27 | leifmadsen | newtonr: actually that might be more appropriate for you ^^^^ |
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| 12:47.27 | pietro | leifmadsen: I'm not the author of the patch, but tis issue is very annoying: TLS is broken with some UA. |
| 12:48.08 | leifmadsen | pietro: which is why I asked some people who can get Digium resources to look at it to make sure the prioritization is correct. Otherwise you'll need to convince someone from the community to steer the issue to completion |
| 12:51.29 | pietro | leifmadsen: great, many thanks. |
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| 13:05.17 | wdoekes | pietro: try https://reviewboard.asterisk.org/r/2123/diff/#index_header |
| 13:05.36 | wdoekes | that's very related |
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| 13:12.32 | pietro | wdoekes: thanks, I'm not expert in Asterisk code flow: how must I do after patch testing ? |
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| 13:14.10 | wdoekes | after testing? you could test the patch, by cd'ing into your asterisk source dir, and run: wget 'https://reviewboard.asterisk.org/r/2123/diff/raw/' -qO- | patch -p3 |
| 13:15.08 | jeffspeff | So, I've got realtime sip working, and I've enabled rtcachefriends... however, voicemail MWI seems to have stopped working |
| 13:17.47 | roe | I hate that MWI is a 'feature' |
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| 13:27.48 | pietro | wdoekes: yes, sure after applying the patch I must test it :) |
| 13:28.12 | wdoekes | you're the one suffering the issue, right? |
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| 13:37.07 | malcolmd | *looks* |
| 13:37.44 | malcolmd | leifmadsen: yup, it's set properly |
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| 13:39.32 | pietro | wdoekes: I'm dealing with this issue: https://issues.asterisk.org/jira/browse/ASTERISK-19003 that seems related to https://issues.asterisk.org/jira/browse/ASTERISK-18345 |
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| 13:40.47 | WIMPy | Has issues.asterisk.org had some upgrade relating TLS? |
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| 13:56.19 | zamba | i have a problem dialing between two devices connected to the same asterisk, but from two different, private networks.. |
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| 13:56.37 | zamba | i see that asterisk tries to bridge them remotely, but i guess this is the core of the problem |
| 13:56.57 | zamba | i've set nat=yes for both peers |
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| 13:57.20 | AkkerKid | Good morning/afternoon/evening/night everyone! |
| 13:58.32 | AkkerKid | Wow... Even the guys who wrote the book are in here. I must have come to the right place! |
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| 14:01.10 | AkkerKid | I'm wondering if it's possible to create a dialplan where queues are generated on the fly from database records. |
| 14:01.50 | WIMPy | zamba: Disable directmedia for those peers. |
| 14:02.30 | AkkerKid | Or should I give up and just put the arugements of the Dial() application into a MySQL record? |
| 14:03.36 | zamba | WIMPy: directmedia=nonat? |
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| 14:15.43 | WIMPy | zamba: That should do it automatically. |
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| 14:32.44 | kresp0 | hi all |
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| 14:34.04 | d_preston215 | Is there a way to start a call muted or start a ChanSpy whipser session muted on the whispering person's end? |
| 14:35.10 | kresp0 | Here I cant load MeetMe because some "/usr/lib/asterisk/modules/meetme.so" file is missing. I'm using asterisk.org repos for debian. |
| 14:35.33 | kresp0 | googled it, but no luck |
| 14:35.35 | AkkerKid | check in your lib64 directory too |
| 14:35.51 | kresp0 | locate meetme.so gives no results :S |
| 14:36.25 | AkkerKid | locate's index isn't in real time so if it's only recently that you've added meetme, it may not be in it yet |
| 14:36.35 | kresp0 | updatedb before |
| 14:36.38 | WIMPy | kresp0: Maybe it's in the dahdi package? |
| 14:36.46 | AkkerKid | a find command might be more thurough |
| 14:37.03 | AkkerKid | righto |
| 14:38.41 | AkkerKid | shouldn't it be called "app_meetme.so"? |
| 14:38.53 | kresp0 | thank you WIMPy, and AkkerKid. dahdi package installed, but still asterisk cant load it |
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| 14:39.10 | WIMPy | Do you have it now? |
| 14:39.27 | kresp0 | still dont WIMPy :/ |
| 14:39.32 | kresp0 | yes AkkerKid |
| 14:39.48 | WIMPy | Then I have no idea where it has been hidden. |
| 14:40.10 | WIMPy | But if you're using Asterisk 10 or 11 ConfBridge might be the better choice anyway. |
| 14:40.18 | AkkerKid | if asterisk wasn't compiled with it, you may have to recompile |
| 14:40.32 | kresp0 | I'm trying this: "module load app_meetme" and this is what I get: http://pastebin.com/raw.php?i=XSF1EXfC |
| 14:41.27 | WIMPy | looks tensely at his browser. |
| 14:41.31 | AkkerKid | so the file still doesn't exist or could it have a permissions issue? |
| 14:42.11 | kresp0 | Still doesn't exist |
| 14:42.49 | kresp0 | Would be a good idea to just copy that file from another i686 machine? |
| 14:43.17 | WIMPy | Only if it's the exact same version. |
| 14:43.49 | kresp0 | ok, ok I see the problem :D |
| 14:44.16 | kresp0 | I installed dadhi packet, but that file is on asterisk-dahdi packed |
| 14:44.24 | kresp0 | facepalm |
| 14:44.29 | AkkerKid | nice |
| 14:44.31 | kresp0 | thank you guys |
| 14:44.37 | AkkerKid | http://ofps.oreilly.com/titles/9780596517342/asterisk-DeviceStates.html |
| 14:44.55 | AkkerKid | 2/3rds of the way down is what i was referencing |
| 14:44.55 | WIMPy | Well, the source comes in one... |
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| 15:06.28 | kresp0 | what is the name of the dadhi module for the linux kernel? |
| 15:07.08 | _Corey_ | I'm having a pretty wicked DPMA issue this morning on a new turn-up... |
| 15:07.14 | Maliuta | kresp0: there are several, depends on your hardware |
| 15:07.15 | kresp0 | Already have the dahdi and asterisk-dahdi packages installed. Tried moprobe dahdi, dummy_dahdi |
| 15:07.43 | kresp0 | Thanks Maliuta. I only need the driver for the MeeMe app |
| 15:07.44 | _Corey_ | anyone seen Asterisk core dump infinitely with a DPMA issue? |
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| 15:08.47 | _Corey_ | I should say, I believe it's res_digium_phone.so based on the backtrace, but I'm not sure... |
| 15:09.01 | _Corey_ | Anyone seen something like this? http://pastebin.com/xkYWSjG1 |
| 15:10.26 | Maliuta | kresp0: you mean meetme? |
| 15:10.42 | kresp0 | err yes :) |
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| 15:11.55 | Maliuta | kresp0: so you probably want dahdi_dummy ... it's a dummy timer for those without hardware |
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| 15:12.17 | WIMPy | If it's a half way recent version, there's no seperate dahdi_dummy any more. Just modprobe dahdi is all. |
| 15:12.34 | Maliuta | WIMPy: define "recent" |
| 15:13.14 | kresp0 | version: 1.8 |
| 15:13.14 | kresp0 | FATAL: Module dahdi_dummy not found. |
| 15:13.14 | kresp0 | FATAL: Module dahdi not found. |
| 15:13.29 | WIMPy | I'm bad with long times, but I'm sure it was not removed this year. |
| 15:14.12 | navaismo | kresp0, how do you install dahdi and for what kernel? |
| 15:14.27 | kresp0 | apt-get install dahdi asterisk-dahdi |
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| 15:14.38 | kresp0 | 2.6.32-5-686 |
| 15:14.48 | Maliuta | kresp0: show me dpkg -l | grep dahdi |
| 15:15.07 | kresp0 | ii asterisk-dahdi 1:1.8.11.1-1digium1~squeeze DAHDI devices support for the Asterisk PBX |
| 15:15.08 | kresp0 | ii dahdi 1:2.2.1.1-1 utilities for using the DAHDI kernel modules |
| 15:15.08 | kresp0 | ii dahdi-linux 1:2.3.0.1+dfsg-2 DAHDI telephony interface - Linux userspace parts |
| 15:15.15 | Maliuta | kresp0: oh, and get a newer kernel ... 2.6 is sooooo 10 years ago ;) |
| 15:16.31 | kresp0 | Maliuta: you mean compiling the kernel by myself after every security update? |
| 15:16.37 | Maliuta | kresp0: and you did m-a dahdi ?? |
| 15:16.44 | kresp0 | mmm. no |
| 15:16.53 | kresp0 | didnt know what is that |
| 15:16.53 | Maliuta | kresp0: I don't use distro kernels |
| 15:17.03 | navaismo | kresp0, I never used repository packages, but if in the directory of asterisk modules exist the chan_dahdi.so try to load and see the errors |
| 15:17.06 | Maliuta | kresp0: you need to have module-assitant install |
| 15:17.31 | Maliuta | kresp0: and the headers/sources for your kernel. |
| 15:17.47 | Maliuta | kresp0: then you need to run "m-a a-i dahdi" as root |
| 15:17.59 | Maliuta | kresp0: it will build and install the modules |
| 15:18.10 | kresp0 | ok, trying that Maliuta |
| 15:18.28 | Maliuta | kresp0: the install to /lib/modules/<kernel_version/dahdi |
| 15:18.57 | Maliuta | I have been meaning to do the work to include a dkms package |
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| 15:19.54 | kresp0 | Maliuta: the modules build and install failed |
| 15:20.03 | kresp0 | the log, says nothing :( |
| 15:20.08 | kresp0 | only starting date |
| 15:20.34 | Maliuta | kresp0: you also need to install dahdi_source :) |
| 15:20.40 | kresp0 | ok one sec |
| 15:21.22 | kresp0 | compiling :) |
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| 15:24.57 | kresp0 | dahdi: Telephony Interface Registered on major 196 |
| 15:25.18 | kresp0 | thank you again Maliuta, navaismo & WIMPy ;) |
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| 15:27.33 | Spads | In 1.8.10.1 the "include =>" entries in my extensions.conf seem to have changed behavior, so that no matter what I do, the included sections end up at the very end of my "dialplan show" output. |
| 15:27.58 | Spads | I am hoping with all my might that this is a big FAQ, but the word "include" is really hard to search for. |
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| 15:31.34 | mattp | when I do a dialplan show <number>@<context> I get two entries for certain numbers. How do I know which one take precedence? |
| 15:32.22 | *** join/#asterisk smooth_penguin (~jr@59.162.86.168) |
| 15:33.38 | mattp | The number, in case it matters, is 112@desk_phones; the results are for '_1.' and '112' |
| 15:34.28 | Maliuta | mattp: I have generally found that the higher one matches first (I'm pretty sure it works as a drop down list) |
| 15:35.05 | Maliuta | mattp: in my case I had to put _1300. above _13XXXX |
| 15:35.34 | mattp | oh, higher in the diaplan file rather than the output? |
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| 15:37.57 | WIMPy | Includes are always serached after all extensions. |
| 15:38.09 | WIMPy | And order in the file does not matter at all. |
| 15:38.23 | WIMPy | The most specific entry will be used. |
| 15:38.35 | Spads | WIMPy: ah, that was not always the case. |
| 15:38.47 | mattp | <PROTECTED> |
| 15:39.07 | WIMPy | When was not always? Asterisk 0.something? |
| 15:39.08 | Spads | broke his _X. type rules out into another included context |
| 15:39.27 | Spads | WIMPy: the version that came with Ubuntu 10.04 didn't do it this way. Just broke in 12.04 |
| 15:39.32 | Spads | shrugs |
| 15:39.42 | mattp | I suppose the core of my question is why are two shown in the first place? From reading the asterisk book, I infer that the strongest match is what applies. |
| 15:39.48 | Spads | you have to dig in the docs pretty deep to figure out why your dialplans suddenly broke |
| 15:39.55 | WIMPy | I'm sure it has not changed in the last 5 years. |
| 15:40.07 | Spads | Curiouser. |
| 15:40.43 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 15:41.08 | jeffspeff | Spads, why not build the latest version; instead of installing packages? |
| 15:41.40 | Spads | jeffspeff: because this is a PBX, not a new game that just came out. |
| 15:41.49 | Spads | it needs to be solid and reliable and supported, not new and flashy and cool |
| 15:42.10 | smooth_penguin | has anyone here thought of forwarding calls from thier android phone over sip |
| 15:42.17 | smooth_penguin | is that possible? |
| 15:42.18 | Spads | anyway thanks for the context WIMPy |
| 15:42.23 | jeffspeff | Spads, and installing whatever package that comes with Ubuntu makes it more stable how? |
| 15:43.01 | Spads | jeffspeff: security updates come in for five years, and that's it. I don't have to go through this hair-pulling about dialplan semantic shifts more than once every five years |
| 15:43.04 | [TK]D-Fender | Include order counts. Within a context does not. |
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| 15:43.28 | jeffspeff | lol |
| 15:43.37 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:43.39 | Spads | jeffspeff: if there were a bug in our version holding back work, and the new version fixed it, we'd upgrade. |
| 15:43.50 | Maliuta | smooth_penguin: I use iaxagent |
| 15:44.14 | Maliuta | smooth_penguin: it works well, and is better with firewalls |
| 15:44.28 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 15:44.37 | WIMPy | Spads: Are you sure Asterisk is the right choice then? |
| 15:45.02 | Spads | WIMPy: are you recommending something more stable? |
| 15:45.22 | WIMPy | No, I can't. |
| 15:45.28 | Spads | WIMPy: Right, well thanks for that. |
| 15:45.43 | smooth_penguin | Maliuta: oh hrm |
| 15:45.47 | Maliuta | smooth_penguin: although SIP works too, there are a couple of apps that do that. I just find IAX more reliable |
| 15:46.18 | Maliuta | smooth_penguin: single TCP port vs 5060+random RTP ports all on UDP |
| 15:46.38 | *** part/#asterisk Spads (~spacehobo@unaffiliated/spads) |
| 15:47.07 | Maliuta | smooth_penguin: from a firewall perspective you need to think not only of your firwewall, but that of you mobile carrier |
| 15:47.11 | smooth_penguin | Maliuta: oh I think I wasnt clear with my question |
| 15:47.17 | mattp | WIMPy: could it be that both strings '112' and '_1.' are 3 chars long ? I've now moved the rules into two different contexts, included by a third common parent and still get two results from dialplan show. |
| 15:47.22 | smooth_penguin | I was wondering if gsm calls recieved on the android phone |
| 15:47.40 | smooth_penguin | can be forwarded over sip to another box or such |
| 15:48.54 | WIMPy | mattp: Sort them literally and you know which order they are used. |
| 15:49.30 | WIMPy | And to sum up extensions inside the context come first only if there is no match at all, the includes are searched in listed order. |
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| 15:50.33 | WIMPy | So a better macht in an includes context will not preceede a more general one in the context itself. |
| 15:51.02 | mattp | ah, that's clearer, thank you. |
| 15:51.04 | jeffspeff | mattp, you probably need more contexts to be more descriptive in how the call flows |
| 15:51.09 | *** join/#asterisk MLNoah (~MLNoah@noc.metalink.net) |
| 15:51.57 | WIMPy | needs to learn writing again :-( |
| 15:52.05 | MLNoah | i'm having some difficulty with QUEUE_MEMBER in asterisk 11, can someone help me verify I'm not missing something easy? |
| 15:52.17 | MLNoah | passing this through the dialplan: same => n,Set(QUEUE_MSG=${QUEUE_MEMBER(scheduling,penalty,Local/${INC_ID}@internal-no-vm)}) |
| 15:52.34 | MLNoah | verified via verbose that the Local/${INC_ID} is expanding out to the expected extension |
| 15:52.47 | MLNoah | getting this error in console: [Sep 20 11:51:32] ERROR[20911][C-00000007]: app_queue.c:7124 queue_function_mem_read: Unknown option penalty provided to QUEUE_MEMBER, valid values are: logged, free, ready, count, penalty, paused, ringinuse |
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| 15:55.02 | MLNoah | for what it's worth, the queue i'm checking, the user isn't actually logged into at the moment. |
| 15:55.15 | mjordan | MLNoah: that's a pretty bad error message, since it implies that it didn't know what the option is |
| 15:55.28 | mjordan | MLNoah: and that is actually the problem. If the interface doesn't exist yet, it won't process the penalty |
| 15:56.01 | MLNoah | i also get the same error if i try to use that for a queue/interface combo that does exist |
| 15:56.05 | mjordan | hm |
| 15:56.07 | mjordan | now that is odd |
| 15:56.16 | MLNoah | trying to read something out to tell me if a user exists in a given queue |
| 15:56.35 | mjordan | it should be looking up the queue member that you passed into the function |
| 15:56.36 | MLNoah | (long term goal: iterate through my queues, find out if user is in any of them, and prompt user via XMPP to see if they want to log out, since they're clocking out) |
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| 15:57.37 | mjordan | MLNoah: are you working off of a fresh checkout of 11? |
| 15:57.44 | MLNoah | oops. sorry. I do not get the error if I call that for a queue/interface that does exist |
| 15:57.48 | MLNoah | check twice, report once >< |
| 15:57.51 | mjordan | :-P |
| 15:57.52 | mjordan | whew |
| 15:57.53 | MLNoah | yes, fresh checkout as of last night |
| 15:58.05 | MLNoah | Asterisk SVN-branch-11-r373133 |
| 15:58.31 | mjordan | MLNoah: kk. I imagine that its probably just a misleading error message - it should probably tell you "hey, there's no queue member for this so I can't apply this option" |
| 15:59.03 | MLNoah | the other issue I see then is that QUEUE_MEMBER is returning 0 if the queue/interface doesn't exist, and also returning 0 if the queue/interface exists at penalty 0 |
| 15:59.41 | MLNoah | and it looks like paused and ringinuse behave that way as well |
| 15:59.57 | MLNoah | is there any other way to tell by the dialplan if queue/interface exists, without removing them from the queue? |
| 16:02.32 | mattp | WIMPy: ah, I've worked out where I was going wrong. dialplan show gives all the /possible/ matches, in order of priority. Not, as I initially assumed, all of the matches that will actually be used. |
| 16:05.03 | mjordan | MLNoah: for queue existence QUEUE_EXISTS. For members, probably QUEUE_MEMBER_LIST. I think the intent of QUEUE_MEMBER was to get information about members in a queue or about a known queue member, but not necessarily to test the existence of a Queue member. |
| 16:05.14 | *** join/#asterisk Sicelo (~sicelo@unaffiliated/sicelo) |
| 16:05.40 | mjordan | it wouldn't be very hard to add an 'exists' specifier to QUEUE_EXISTS |
| 16:05.47 | mjordan | er, QUEUE_MEMBER that is |
| 16:05.59 | mjordan | just to test if a queue member actually exists in that queue or not |
| 16:06.24 | leifmadsen | nice |
| 16:06.38 | leifmadsen | doable in dialplan through a loop, but that's kind of messy |
| 16:07.14 | leifmadsen | something that returned 0 or 1 in QUEUE_MEMBER though would be nice |
| 16:08.09 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
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| 16:15.01 | AkkerKid | Anyone know if queues can be created on the fly within the dialplan based on results from a MySQL query? |
| 16:16.48 | AkkerKid | I'd like to be able to query a database for extensions an incoming call should go to but not predefine the queue as they'll be changing quite a bit as my company grows. |
| 16:18.03 | AkkerKid | My "Definitive Guide" only goes so far. :) |
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| 16:20.15 | MLNoah | consider me +1 for adding exists to QUEUE_MEMBER |
| 16:21.00 | MLNoah | the other thing I thought of as a short term method would be to wrap my set in a TryExec, since that should kick back a fail if the Set errors out because the interface isn't a member of the queue? |
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| 16:25.04 | cneb3000 | come back to the asterisk community for the first time in 1 1/2 years and see you're 'tentatively' close to releasing v11. |
| 16:25.05 | volga629 | what is mean ccss.c:906 ast_set_ccnr_available_timer: 0 is an invalid value for ccnr_available_timer. Retaining value as 7200 |
| 16:25.08 | cneb3000 | first world problems ¬_¬ |
| 16:27.15 | *** join/#asterisk sustav (~vpp@nat/digium/x-agjmprvreuomzgky) |
| 16:27.29 | cneb3000 | volga629: what do you do to get that error? |
| 16:28.51 | volga629 | when I am testing call from vpn tunnel to asterisk |
| 16:29.31 | volga629 | also I am get this chan_sip.c:9147 process_sdp: ignoring 'video' media offer because port number is zero |
| 16:29.53 | cneb3000 | so an external phone (that isn't registered to astrisk) is phoning a phone that is registered? |
| 16:30.17 | cneb3000 | do you know what wireshark is? |
| 16:30.19 | *** join/#asterisk evil-man (~evil-man@insider-mail.icf.org.ru) |
| 16:30.33 | cneb3000 | is SOUNDS like whatever is calling the asterisk server probably isn't formatting the packets right..? maybe? |
| 16:30.44 | cneb3000 | you see the error 'ignoring 'video' media offer because port number is zero' |
| 16:30.53 | cneb3000 | the important part is probably the port being 0 :) |
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| 16:31.42 | cneb3000 | volga629: can you explain the call in more detail? for example |
| 16:31.55 | volga629 | Yes, the right 0, currently h264 |
| 16:32.15 | cneb3000 | im no expert, but I dont think you can give something port 0 |
| 16:32.46 | cneb3000 | so if you say in a SIP/SDP packet to return media to 'port 0' it probably wont work? |
| 16:33.01 | cneb3000 | im no expert, just sharing ideas :) |
| 16:33.04 | volga629 | I have remote branch vpn tunnel and another end video conference phone |
| 16:33.34 | cneb3000 | are you trying to do a video call? |
| 16:33.43 | cneb3000 | i.e. with voice and video? |
| 16:33.52 | volga629 | yes it will not work with 0, but I don't know where exactly to start trouble shoot because voice only without video work perfect |
| 16:34.06 | cneb3000 | ahh ok - what kind of phone is it? |
| 16:34.16 | volga629 | fanvil se 780 |
| 16:34.22 | cneb3000 | there's probably a setting in there to define which port to use for video |
| 16:34.34 | volga629 | it connected to TV through HDMI |
| 16:35.07 | MLNoah | mjordan/leifmadsen: apparently TryExec isn't going to let me emulate an exists functionality for QUEUE_MEMBER as i'm currently doing it... returning Success whether or not the interface exists in the queue. any other suggestions on what I might try? |
| 16:35.32 | cneb3000 | volga629: I cant find an english manual :( but there's probably a setting on the phone that you use to say 'use port x for video and port x for voice' |
| 16:35.35 | leifmadsen | MLNoah: you'll have to show me an example of what you're trying to do |
| 16:35.52 | MLNoah | same => n,TryExec(Set(Q_MSG=${QUEUE_MEMBER(scheduling,penalty,Local/${INC_ID}@internal-no-vm)})) |
| 16:35.53 | volga629 | I have locally here few phones and they working no issue ( both video and audio) |
| 16:36.08 | cneb3000 | volga629: are they the same type of phone? |
| 16:36.15 | leifmadsen | I don't think you are using TryExec() correctly |
| 16:36.17 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 16:36.26 | MLNoah | hrm, ok. |
| 16:36.29 | leifmadsen | explain what you're trying to do |
| 16:36.52 | volga629 | http://www.fanvil.com/List.asp?C-1-66.html |
| 16:37.04 | volga629 | yes all the same |
| 16:37.29 | cneb3000 | volga629: don't complicate things. just check the configuration on each phone the same |
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| 16:37.54 | MLNoah | I want to determine if the Local/... interface exists in each of my queues. I get an error on the console if the member doesn't exist in the queue, but Q_MSG is getting set to 0 if the member doesn't exist |
| 16:38.11 | volga629 | I did but phone behind tunnel and it not same |
| 16:38.23 | MLNoah | (though I guess i could change so all of my members go in at penalty 1 instead of penalty 0, which would allow me to distinguish penalty 0 from not-a-member 0. duh.) |
| 16:38.35 | cneb3000 | volga629: so this is the only phone behind a tunnel? |
| 16:38.58 | MLNoah | anyway, the original thought behind tryexec was to catch the QUEUE_MEMBER error and be able to flag whether it was actually returning 0 or was erroring out on a non-existing member and throwing 0. |
| 16:39.05 | as001 | Hi I am struggling to pass callerid from my atcom ip02 Asterisk device to my Dahdi/1 (panasonic cordless phone). I can see caller id on my atcom ip02 box (on cli) but when I do dial(Dahdi/1) I can see just Incoming on phone like I did not send callerid. Can you help me with this issue ? |
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| 16:39.49 | cneb3000 | volga629: could it be the tunnel is adjusting the port to 0 for some reason? |
| 16:41.19 | volga629 | Let try use tcpdump |
| 16:41.32 | cneb3000 | good idea |
| 16:41.42 | cneb3000 | if you can do it before the tunnel and after the tunnel |
| 16:41.50 | cneb3000 | we can see if data is being changed! |
| 16:44.15 | cneb3000 | holy crap just reading latest on asterisk (i stopped using for a while just as 1.8 came out). |
| 16:44.20 | cneb3000 | asterisk 10 load tests :| |
| 16:44.29 | cneb3000 | probably old news to some of you |
| 16:44.35 | cneb3000 | applauds dev team |
| 16:48.25 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 16:48.44 | leifmadsen | load tests? |
| 16:48.58 | cneb3000 | ugh, not official. just someones comments on voip-info |
| 16:49.30 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:49.55 | cneb3000 | "I had done some load tests with Asterisk 10 and my highest results was: 1750 calls per seconds (cps) up to 13000 concurrent calls done on a intel xeon with dual six core and hyperthreading (= 24 cores) and 12 GB ram. the sysload was around 2.5 during this test (with open rtp ports but without much" |
| 16:50.01 | cneb3000 | - http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 16:51.55 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 16:54.01 | volga629 | http://fpaste.org/wRwA/ that what I see |
| 16:56.49 | volga629 | I will come back in 10 min need restart |
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| 17:13.28 | navaismo | can i add a custom field in the cdr and then in the dialplan use --> Set(CDR(myfield)=${EXTEN}_${UNIQUEID}_${MYVAR}) |
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| 17:20.50 | navaismo | because is not working... |
| 17:22.11 | cneb3000 | navaismo: have you tried the example out the book? |
| 17:23.50 | cneb3000 | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-SysAdmin-SECT-1 |
| 17:24.47 | Qwell | "the cdr"? |
| 17:24.56 | leifmadsen | navaismo: only if you use cdr_adaptive_odbc |
| 17:25.02 | Qwell | or csv I think |
| 17:25.42 | Qwell | rather, cdr_custom? whichever one it is. |
| 17:25.50 | Qwell | the one that slightly different than the one it's exactly the same as |
| 17:26.27 | navaismo | ah ok, thanks. |
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| 17:38.17 | nickgaw | Hi, Anyone here use pbx in a flash? I am trying to find out if the flash drive iso image installs the same stuff as the normal CD iso image? |
| 17:40.04 | leifmadsen | #pbxinaflash |
| 17:40.34 | nickgaw | ok did not know that channel existted. |
| 17:40.40 | *** part/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 17:40.49 | leifmadsen | only assumes it exists |
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| 17:45.06 | drmessano | PBX Pilaf, blah |
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| 18:02.08 | navaismo | this guy: nickgaw claim to be totally blind and if so its amaizing!! |
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| 19:41.21 | MLNoah | what's the best place to submit/discuss something that probably isn't strictly speaking a bug, but appears to be sub-optimal handling? specifically, I'm working on an interactive XMPP interface for my timeclock and queues, but sendtodialplan is sending both unsolicited and solicited messages through the incoming context handling. I would expect that messages that were "caught" by a |
| 19:41.21 | MLNoah | JABBER_RECEIVE solicited method shouldn't ever trip sendtodialplan. |
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| 19:47.22 | mjordan | MLNoah: hm. |
| 19:47.39 | mjordan | MLNoah: what is your interface written in? |
| 19:47.43 | MLNoah | dialplan |
| 19:47.54 | mjordan | k. You might still be best off on the -users list |
| 19:48.07 | MLNoah | ok, thanks |
| 19:49.49 | AkkerKid | Does anyone know if there are going to be mid-level dialplan classes at AstriCon in October? |
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| 21:02.37 | gavimobile | is it normal that asterisk is using Dial several times before executing a call? |
| 21:05.31 | *** join/#asterisk carrar (tim@osburn.com) |
| 21:05.39 | carrar | w00t |
| 21:05.59 | carrar | Did I miss anything? |
| 21:07.00 | ChannelZ | gavimobile: no.. asterisk does whatever you tell it to do |
| 21:07.35 | gavimobile | ChannelZ: I hope you had a good sleep:-p |
| 21:07.40 | carrar | asterisk: Make me a sammich |
| 21:07.44 | ChannelZ | carrar: yes.. Katty handed out cookies. |
| 21:07.49 | carrar | WHAT! |
| 21:07.55 | carrar | I knew I missed something good |
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| 21:24.43 | gavimobile | ChannelZ: your answer is logical, what about my sip provider? |
| 21:24.54 | gavimobile | my itsp* |
| 21:25.18 | gavimobile | could they be the problem which causes my pbx to attemplt several times before the call gets connected? |
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| 21:29.43 | gavimobile | nevermind, im going to re-read over the section Pattern-matching examples in the manual |
| 21:41.21 | gavimobile | nope.. I already knew this |
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| 21:49.33 | ChannelZ | We're not clairvoyant and don't even really know what you're talking about. Pastebin some verbose console output. |
| 21:49.53 | ChannelZ | Asterisk will not 'dial multiple times until success' by its self. |
| 21:51.07 | jpsharp | Haunted Asterisk |
| 21:51.15 | jpsharp | The calls are coming from inside the server! |
| 21:52.48 | ChannelZ | shits a brick |
| 21:53.14 | specing | calls the masons |
| 21:54.06 | gavimobile | ChannelZ: glad you asked http://pastebin.com/JEDSt4WU |
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| 22:00.54 | ChannelZ | Looks more like you have a loop or something. |
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| 22:01.35 | gavimobile | ChannelZ: and during the call it continues to loop |
| 22:02.01 | albania-asterisk | hello there i need somebody opinion for heavy asterisk systems |
| 22:02.03 | gavimobile | my extentions.conf file has like 4 lines in it no cause I removed anything and everything which wasn't necessary to this example |
| 22:02.19 | albania-asterisk | our system has more than 1000 concurrent calls usually |
| 22:02.24 | ChannelZ | pastebin your entire extensions.conf and sip.conf minus passwords |
| 22:02.25 | gavimobile | extensions.conf |
| 22:02.34 | gavimobile | ChannelZ: k |
| 22:02.51 | albania-asterisk | and we have udp errors and when monitored via other sources, there 5-15% packet loss |
| 22:02.57 | albania-asterisk | on UDP ping |
| 22:03.20 | albania-asterisk | i believe is not related to network bandwidth as on icmp ping it has 0% packet loss |
| 22:03.40 | albania-asterisk | can it be related to udp conf on system which is centos 64 bit |
| 22:03.42 | ChannelZ | albania-asterisk: are these 1000 calls on a local network or over the internet? |
| 22:03.54 | albania-asterisk | over the internet |
| 22:04.20 | albania-asterisk | when it goes to 1400-1500 concurrent calls the udp packet loss starts to be stable on 15% |
| 22:04.51 | albania-asterisk | every 1 or 2 sec , 10 samples are sent for monitoring udp packet loss |
| 22:06.12 | gavimobile | ChannelZ: sip.conf http://pastebin.com/XqcA0DbU |
| 22:07.26 | gavimobile | ChannelZ: extensions.conf (no I didn't cut anything.. I did the test with my extensions file like this)http://pastebin.com/TfFw8SQ0 |
| 22:07.31 | gavimobile | http://pastebin.com/TfFw8SQ0 |
| 22:08.32 | gavimobile | im starting to think ram... |
| 22:09.00 | albania-asterisk | any idea ? |
| 22:09.21 | ChannelZ | Well assyuming your asterisk server isn't running out of resources and is the one dropping packets, maybe your upstream provider is having issues with the amount of traffic. |
| 22:09.29 | ChannelZ | gavimobile: are you calling yourself as a test? |
| 22:10.18 | volga629 | would syntax for wget to apply asterisk patch ? |
| 22:10.34 | volga629 | -O not remember |
| 22:10.46 | albania-asterisk | upstream provider does not have issues , as other type of traffic does not have packet loss as there is no congestion |
| 22:11.21 | ChannelZ | albania-asterisk: I mean your ITSP |
| 22:11.47 | gavimobile | ChannelZ: no... cellphone |
| 22:11.58 | gavimobile | I didn't get to outgoing yet... |
| 22:12.14 | gavimobile | I am NOT calling myself as the test |
| 22:12.47 | albania-asterisk | it is not iTSP as it big TIER1 and further more we see same things from different sources when monitoring udp packet loss |
| 22:13.33 | albania-asterisk | i thought as the bandwidth goes 150-200 mbps during this time |
| 22:13.50 | gavimobile | Mbps? |
| 22:14.09 | gavimobile | megabites |
| 22:14.10 | albania-asterisk | the amount of packets per sec is huge and OS may neesd some tunning to handle better buffering or other things |
| 22:14.18 | albania-asterisk | mbps |
| 22:14.32 | volga629 | How to apply asterisk-10.5.1-patch.gz |
| 22:14.42 | ChannelZ | gavimobile: but this number that you're dialing, 0773354673, is a separate cell, not your own DID? |
| 22:15.06 | gavimobile | 077 is a land line |
| 22:15.16 | gavimobile | its my main incoming did |
| 22:15.29 | gavimobile | a land line number* |
| 22:15.29 | ChannelZ | yeah... so you're calling yourself. |
| 22:15.45 | gavimobile | in that sense, yes im calling myself |
| 22:16.12 | gavimobile | excuese my misunderstanding |
| 22:16.48 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 22:18.10 | albania-asterisk | are you aware of this problems coming thourgh on these amount of concurrent calls ?! |
| 22:19.22 | kuku | I'm trying to do chanspy with just using the sip extension ( without the channel ) it works 80% of the time, the other 10% it doesn't and the other 10% it connects to an entirely different extension... any advice ? |
| 22:19.33 | ChannelZ | gavimobile: Well your problem is that you are calling in a loop. You call out, which spawns a call in. However your "netvision-trunk" peer where the call is supposed to come in (I assume) doesn't match that peer because the IP addresss are different. |
| 22:19.34 | navaismo | volga629, usually a patch is applied with: patch -p1 < patchfile If the patch cant find the file to patch it will ask you |
| 22:20.46 | ChannelZ | So it's treating the call as anonymous/unauthenticated which in turn goes to your 'motekPC' context in your dialplan, which then dials out again. |
| 22:21.42 | albania-asterisk | are you aware of this problems coming thourgh on these amount of concurrent calls ?! |
| 22:22.24 | ChannelZ | albania-asterisk: I'm not personally aware of a problem with an arbitrary number of calls but can see how a large number could cause problems. If you have a crappy network card or a slow CPU which is not getting through all of the incoming packets fast enough it's overflowing a buffer and dropping... remember UDP is "best effort" |
| 22:22.59 | ChannelZ | What codec are you using? |
| 22:23.13 | gavimobile | ChannelZ: yes I call imiditly after the pbx answers the call. and it ususally happenes after the 3rd or 4th time.. will it be problematic later on? |
| 22:23.20 | volga629 | I rember on mantis was possible use wget -O, but file is compressed to .gz |
| 22:23.33 | ChannelZ | gavimobile: it's problematic NOW because you're creating a call loop. |
| 22:23.54 | gavimobile | about netvision-trunk. the ip I provided forwards apparently to a diff ip.. the ip in my sip.conf is the ip they provide to their clients |
| 22:24.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:24.36 | gavimobile | ChannelZ: will the same thing happen if my pbx gets 3-4 calls in a short range of time from different callers? |
| 22:25.02 | albania-asterisk | channelz: the server is really a beast, dual xeon with 1333 bus, 16 cores, 16 gb of RAM, and network card is 1gbps to gigabit swithc port as well, so do you have experience with asteirsk servers handling more than 1700 concurrent calls with normal quality of voice, most of call are p2p, 5% can be with transcoding from g711 to g729 |
| 22:25.40 | ChannelZ | Well you can see from your console output that a call is coming in from 82.166.66.43 which is not the same as 82.166.67.43 as specified in your sip.conf, so it doesn't match that peer and winds up in the wrong context which in turn dials out again thinking it's you making another outgoing call. |
| 22:26.22 | volga629 | remember |
| 22:27.01 | gavimobile | im leaving them soon, but I need them for the 077 number.. can I add two hosts in sip.conf for the netvision-trunk context |
| 22:27.20 | ChannelZ | If your ITSP sends you calls from multiple IPs then you have to define the peer for them a different way such that it matches by From or something. |
| 22:28.12 | WIMPy | Just that ITSPs usually don't authenticate. |
| 22:28.16 | elguero | volga629: wget -O - [diff-url] | patch -p0 |
| 22:28.56 | *** join/#asterisk Neptu (~Neptu@m176-71-247-192.cust.tele2.se) |
| 22:28.58 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-235.west.biz.rr.com) |
| 22:29.11 | gavimobile | ChannelZ: that I don't know, but what I do know is I was using the same host that I was always using on my previous freepbx machine which I am retiring now for the last year or so... never had any problems with inbound calls coming in |
| 22:29.42 | gavimobile | on the other hand I didn't know how to check the cli or logs to see if this was happening |
| 22:30.01 | gavimobile | maybe my last statement wasn't very relavent |
| 22:31.49 | ChannelZ | Well if you want to live life by "it worked before" in spite of this being a completely different system, and in spite of what I'm seeing in your own console output and configs, well then I can't help you much more. |
| 22:32.02 | volga629 | atch: **** Only garbage was found in the patch input. |
| 22:32.06 | volga629 | patch |
| 22:32.24 | ChannelZ | I told you why it's messed up and doing what it's doing. |
| 22:32.36 | gavimobile | ChannelZ: no I want my new pbx to be the best I can possibly make it |
| 22:32.48 | elguero | volga629: you trying to apply a patch from a Jira issue? what number? what patch? |
| 22:33.03 | WIMPy | gzip -d ? |
| 22:33.17 | gavimobile | how can I find out if they provide me calls on multiple ip's? there support is as good as speaking to a wall |
| 22:33.40 | ChannelZ | The fact is, in the pastes you've provided, your incoming call is such that it turns into another outgoing call to yourself which turns into another incoming call, etc. |
| 22:33.44 | WIMPy | Wait for calls to come? |
| 22:33.53 | albania-asterisk | i have asterisk dual xeon with 1333 bus, 16 cores, 16 gb of RAM, and network card is 1gbps to gigabit swithc port as well, on 1000-1400 concurrent calls i am experiencing increase on UDP packet loss % from 5-20%, but being packet loss only on udp traffic, other type of traffic do not have congestion, so do you have experience with asteirsk servers handling more than 1700 concurrent calls... |
| 22:33.55 | albania-asterisk | ...with normal quality of voice, most of call are p2p, 5% can be with transcoding from g711 to g729 |
| 22:34.00 | WIMPy | oops |
| 22:34.41 | WIMPy | albania-asterisk: Maybe you should increase the buffer on the network card. |
| 22:35.29 | volga629 | first need | to gzip and after pipe to patch ? |
| 22:35.34 | ChannelZ | I'm only guessing that the only reason why this works occasionally is because they DO send you calls on both IPs (maybe more) and you eventually hit one that does match that one specified in the peer, which sends the call to the correct context. |
| 22:36.07 | ChannelZ | But you've probably got massive latency on that "successful" call because it's looping through 2 different providers one or more times. |
| 22:36.38 | albania-asterisk | that is an option, but i need you/someone to tell me that if you are using asterisk on this number of concurrent calls without tuning OS NIC conf or network parameters |
| 22:36.41 | gavimobile | ChannelZ: I see. that would also explain my problem you were trying to assist me with before you went to sleep |
| 22:36.49 | elguero | volga629: I guess I am not understanding... you mentioned what you used to do in Mantis... are you trying to apply a patch from jira (issues.asterisk.org) using wget? |
| 22:37.19 | WIMPy | albania-asterisk: then...? |
| 22:37.39 | gavimobile | ChannelZ: my diagnosis then wasn't the best because I noticed there were a lot of errors in my dialplan.. I cleared that up I believe... |
| 22:38.32 | albania-asterisk | wimpy: so i need to find out that is this about tuning only , or i should mantain an asterisk on this number of concurrent calls for production ? |
| 22:38.32 | volga629 | wget --header="accept-encoding: gzip" -O - http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-10.5.1-patch.gz | gzip -d | patch -p0 |
| 22:38.57 | albania-asterisk | wimpy: i should not mantaina ... |
| 22:39.17 | volga629 | thank you Everyone |
| 22:39.30 | ChannelZ | gavimobile: well the cheap fix is to duplicate the peer in sip.conf with a different name and the other IP so those calls get routed right, and then watch your console to see if even more IPs appear. |
| 22:39.39 | WIMPy | albania-asterisk: I don't really understand that question. Maybe you can re-phrase that? |
| 22:40.13 | WIMPy | Yes, that's fun. I've got 36 peers for one of my ITSPs. |
| 22:40.20 | gavimobile | ChannelZ: hrm... actually on freepbx I had several of the same trunks with different names |
| 22:40.42 | gavimobile | netvision1,2,3 they give me 1 channel per trunk so I created 3 entries |
| 22:41.03 | gavimobile | ChannelZ: ill try that fix.. and diagnose again |
| 22:41.28 | albania-asterisk | wimpy: so i am using server for production and usually it is running on 1100-1500 concurrent calls, now i have this problem of UDP packet loss coming , is it "saying" to me that get another asteirsk box and dont load more the current one , or is matter of just tunning the server ? |
| 22:41.39 | albania-asterisk | wimpy: i dont know if i was clear this time :( |
| 22:41.44 | gavimobile | my logic may very well be wrong, but can my pbx do something in the dialplan to control these loop calls. |
| 22:42.15 | gavimobile | like take the incomming calls and put them on some channels and leave the main channels open for the next incomming call |
| 22:42.39 | ChannelZ | yeah, specify a different context in your [general] section so they go somewhere else |
| 22:43.26 | gavimobile | ChannelZ: ok.. ill try both of these ideas |
| 22:43.45 | gavimobile | thanks! I like you better when its not 4 am your time :-p |
| 22:43.56 | gavimobile | j/k |
| 22:44.02 | ChannelZ | me too |
| 22:44.31 | albania-asterisk | wimpy: did you get me? |
| 22:45.34 | albania-asterisk | wimpy: so i am using server for production and usually it is running on 1100-1500 concurrent calls, now i have this problem of UDP packet loss coming , is it "saying" to me that get another asteirsk box and dont load more the current one , or is matter of just tunning the server ? |
| 22:46.51 | elguero | albania-asterisk: Wouldn't you want to try and tune the server to find out? That is probably the only way to find out if you have hit your capacity for that server or not |
| 22:47.57 | albania-asterisk | elguero: thanks for answering first, ok i dont know how to tune the nic or network paramters for asterisk in these cases |
| 22:48.18 | albania-asterisk | elguero: which are the variables and values to be set and where |
| 22:48.48 | albania-asterisk | elguero: ulitmits are all set optimal, what to tune can be only the nic or os network parameters |
| 22:49.51 | elguero | albania-asterisk: I beleive the suggestion was to try increasing the buffers on the network card if you are positive the packet loss is coming from there |
| 22:50.46 | albania-asterisk | how to change the buffering on the network card on centos ? |
| 22:51.08 | sruffell | albania-asterisk: I would look at the output of "sudo /sbin/sysctl -a | grep ipv4\.udp" Those settings are described here: http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob;f=Documentation/networking/ip-sysctl.txt;h=ca447b35b8333106cdd19649d943d80cdb12cc1e;hb=HEAD#l580 |
| 22:51.28 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 22:52.44 | albania-asterisk | sruffell: output is this: |
| 22:52.45 | WIMPy | Why do ITSPs always do their maintenance during the main work time? |
| 22:52.53 | albania-asterisk | net.ipv4.udp_wmem_min = 4096 |
| 22:52.55 | albania-asterisk | net.ipv4.udp_rmem_min = 4096 |
| 22:52.56 | albania-asterisk | net.ipv4.udp_mem = 2324160 3098880 4648320 |
| 22:53.27 | albania-asterisk | these should be changed in my case ? |
| 22:54.35 | albania-asterisk | what values should i set ? |
| 22:55.01 | sruffell | huh….how much memory do you have on this server? I have 4GB and only show: |
| 22:55.03 | sruffell | # sysctl net.ipv4.udp_mem |
| 22:55.03 | sruffell | net.ipv4.udp_mem = 89760 119682 179520 |
| 22:55.04 | WIMPy | albania-asterisk: Do you see dropped packets in the i/f? |
| 22:55.09 | sruffell | but no…I don't think you should need to adjust those. |
| 22:55.49 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:57.05 | albania-asterisk | i have 16 gb of ram |
| 22:57.15 | sruffell | albania-asterisk: yeah..if you haven't already looked, what is the output of "netstat —interfaces=eth0" |
| 22:57.17 | albania-asterisk | and output of netstat -su |
| 22:57.28 | albania-asterisk | netstat -su |
| 22:57.29 | albania-asterisk | IcmpMsg: |
| 22:57.31 | albania-asterisk | <PROTECTED> |
| 22:57.32 | albania-asterisk | <PROTECTED> |
| 22:57.34 | albania-asterisk | <PROTECTED> |
| 22:57.35 | albania-asterisk | <PROTECTED> |
| 22:57.37 | albania-asterisk | <PROTECTED> |
| 22:57.38 | albania-asterisk | <PROTECTED> |
| 22:57.40 | albania-asterisk | <PROTECTED> |
| 22:57.41 | albania-asterisk | <PROTECTED> |
| 22:57.43 | albania-asterisk | <PROTECTED> |
| 22:57.44 | albania-asterisk | <PROTECTED> |
| 22:57.46 | albania-asterisk | Udp: |
| 22:57.47 | albania-asterisk | <PROTECTED> |
| 22:57.47 | WIMPy | ~pb |
| 22:57.48 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 22:57.49 | albania-asterisk | <PROTECTED> |
| 22:57.50 | albania-asterisk | <PROTECTED> |
| 22:57.52 | albania-asterisk | <PROTECTED> |
| 22:57.53 | albania-asterisk | IpExt: |
| 22:57.55 | albania-asterisk | <PROTECTED> |
| 22:58.58 | albania-asterisk | didnt know that |
| 22:59.04 | albania-asterisk | so link is http://pastebin.com/gF8EZMV5 |
| 22:59.19 | albania-asterisk | the output of netstat -su |
| 22:59.42 | albania-asterisk | you think i should apply this : |
| 22:59.48 | albania-asterisk | # sysctl net.ipv4.udp_mem |
| 22:59.49 | albania-asterisk | sruffellnet.ipv4.udp_mem = 89760 119682 179520 |
| 23:00.05 | sruffell | no…you shouldn't need to touch that. |
| 23:01.26 | sruffell | and with that…I have to run |
| 23:01.42 | albania-asterisk | ok thanks |
| 23:01.47 | albania-asterisk | any other idea ? |
| 23:04.30 | elguero | <sruffell> albania-asterisk: yeah..if you haven't already looked, what is the output of "netstat interfaces=eth0" |
| 23:04.39 | albania-asterisk | and also netstat --interfaces=eth0 |
| 23:04.41 | albania-asterisk | Kernel Interface table :Iface MTU Met RX-OK RX-ERR RX-DRP RX-OVR TX-OK TX-ERR TX-DRP TX-OVR Flg :eth0 1500 0 493410864095 2 0 1988 481481917367 0 0 0 BMRU |
| 23:06.16 | albania-asterisk | errors 2 only on RX |
| 23:08.37 | albania-asterisk | can you ask you guys what it the max concurrent calls you are using asterisk in a producion system ? |
| 23:10.38 | WIMPy | Ok, some overruns. Is I said in the beginning, I'd try to increase the txqueuelen. |
| 23:11.30 | albania-asterisk | ok i can try, how to do that ? |
| 23:12.19 | WIMPy | ip |
| 23:12.27 | WIMPy | or ifoconfig migt also have it. |
| 23:13.50 | albania-asterisk | should i have to install this as it is not there ? |
| 23:14.58 | WIMPy | no. Typo. |
| 23:15.02 | WIMPy | ifconfig |
| 23:16.20 | albania-asterisk | but what is the full command for ifconfig to do this ? |
| 23:17.13 | WIMPy | ip link set <if> txqueuelen <len> |
| 23:19.02 | albania-asterisk | does it need a reboot ? |
| 23:19.13 | WIMPy | No |
| 23:19.20 | WIMPy | Instant effect. |
| 23:21.08 | albania-asterisk | ok i will try tomorrow on peak hours , as not the load is low here |
| 23:21.54 | WIMPy | Do you have some fancy scheduler configured? |
| 23:22.34 | albania-asterisk | what type of scheduler you mean ? |
| 23:22.48 | albania-asterisk | on system , no cron jobs here |
| 23:23.00 | WIMPy | On the interface. |
| 23:23.06 | *** join/#asterisk sbszulu (~Sibusiso_@41.48.192.115) |
| 23:23.18 | albania-asterisk | no nothing , just default thing |
| 23:23.39 | WIMPy | ok |
| 23:23.44 | albania-asterisk | queuelen should be set for tx or rx |
| 23:23.59 | albania-asterisk | maybe on receiving is the problem |
| 23:24.05 | albania-asterisk | or i set it on both ? |
| 23:24.51 | kuku | I'm trying to do chanspy with just using the sip extension ( without the channel ) it works 80% of the time, the other 10% it doesn't and the other 10% it connects to an entirely different extension... any advice ? |
| 23:25.06 | WIMPy | Do you have dropped packets on rx? |
| 23:25.37 | volga629 | Is anybody know about IMAP storage warning like app_voicemail.c:2840 mm_log: IMAP Warning: Unknown message data: 34 FETCH ? |
| 23:25.47 | *** part/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 23:26.19 | albania-asterisk | there are huge number of packets and only 2 packet error on RX and Tx 0 error packets |
| 23:26.58 | WIMPy | Errors are real errors. i.e. broken packets. |
| 23:27.22 | albania-asterisk | when i see udp packets with netstat -su, i have many error udp packets |
| 23:27.29 | WIMPy | Like a broken interface or evil software on the other end or a bad cable. |
| 23:27.49 | WIMPy | No, look at the interface. |
| 23:28.13 | ChannelZ | You had read overruns |
| 23:28.39 | WIMPy | missed that. |
| 23:29.08 | albania-asterisk | see the output http://pastebin.com/HqtfUUtE |
| 23:29.32 | ChannelZ | at least I think I counted right, the formatting was not great pasted here |
| 23:29.56 | WIMPy | can't connect to pastebin.com again. |
| 23:30.17 | ChannelZ | lots-o-RX, 2 RX errors, 0 dropped, 1988 RX overruns |
| 23:30.20 | volga629 | overruns:1988 look like might duplex mismatch |
| 23:30.21 | ChannelZ | He pasted it right in the channel |
| 23:30.43 | ChannelZ | so dead pastebin don't matter :) |
| 23:31.00 | WIMPy | Oh, then I got it wrong. Thought that was under tx. |
| 23:31.12 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 23:31.30 | ChannelZ | No I think those were all 0. |
| 23:32.09 | albania-asterisk | so is receiving |
| 23:32.29 | ChannelZ | Can you even tweak the RX buffer? Wouldn't that be on the adapter its self (I can't imagine the system getting behind) |
| 23:32.33 | albania-asterisk | this link says a couple of things to do in case of overrun |
| 23:32.35 | albania-asterisk | https://supportforums.cisco.com/thread/160577 |
| 23:33.20 | albania-asterisk | 1988 is big number ? |
| 23:33.28 | WIMPy | No, that's not changed on the fly (or at all). |
| 23:33.43 | WIMPy | No. doesn't seem much, but it's not good, either. |
| 23:33.48 | ChannelZ | Not huge based on the total number of packets but it's still stuff that wound up on the floor. |
| 23:34.38 | albania-asterisk | so what should i do increase the rxqueuelength only ? |
| 23:35.30 | WIMPy | get better hardware. |
| 23:35.43 | albania-asterisk | and the udp errors in netstat -su are they also related , but are bigger in number ? |
| 23:35.59 | albania-asterisk | better hardware for nic or whole server ? |
| 23:36.06 | WIMPy | You might be able to decrease th txbuffer to free ram for the rxbufer, but that depends on the driver. |
| 23:36.20 | WIMPy | The NIC. |
| 23:36.35 | WIMPy | Yes, te UDP errors look strange. |
| 23:36.56 | WIMPy | And they might be the real issue. |
| 23:37.06 | WIMPy | But I have no idea where they might come from. |
| 23:37.30 | WIMPy | Maybe wireshark could help to identify what's going on there. |
| 23:37.59 | WIMPy | But I woulndn't want to dig through that pile of packets to try and find the needle in the haystack. |
| 23:38.01 | ChannelZ | does lspci show your ethernet device sharing an interrupt with something else? |
| 23:38.02 | albania-asterisk | but tell me , is normal to have prodcution asterisk server on 1300-1500 concurrent calls ? |
| 23:39.25 | albania-asterisk | what is command for lspci |
| 23:39.42 | WIMPy | giggles |
| 23:40.35 | WIMPy | Do you have a kernel with SMP aware network receive functions? |
| 23:40.36 | ChannelZ | If your actually getting packet loss, it doesn't have much to do with Asterisk specifically. |
| 23:41.04 | ChannelZ | and lspci is just lspci |
| 23:42.03 | albania-asterisk | didn't know :) |
| 23:42.04 | ChannelZ | though it may or may not tell you anything of use |
| 23:42.10 | albania-asterisk | 07:00.0 Ethernet controller: Intel Corporation 82576 Gigabit Network Connection (rev 01) |
| 23:42.12 | albania-asterisk | 07:00.1 Ethernet controller: Intel Corporation 82576 Gigabit Network Connection (rev 01) |
| 23:42.18 | albania-asterisk | this what i have only |
| 23:42.45 | WIMPy | cat /proc/interrupts |
| 23:43.51 | albania-asterisk | is large , putting on pastebin |
| 23:44.28 | albania-asterisk | http://pastebin.com/A3RBuVX1 |
| 23:44.41 | albania-asterisk | or should i put in notepad? |
| 23:45.43 | volga629 | chan_sip.c: ignoring 'video' media offer because port number is zero what would be good start to trouble shoot this error on same extension voice working no isse |
| 23:45.46 | volga629 | issue |
| 23:46.45 | ChannelZ | It looks like they're all separate |
| 23:47.04 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:47.10 | WIMPy | volga629: To me it sounds like a broken SDP offer. But if you've got a sip debug if that, someone knowing more aout it may be able to give a better answer. |
| 23:47.43 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 23:48.10 | WIMPy | Yes, all MSI. |
| 23:48.13 | ChannelZ | I haven't done any video with * but it sounds like the device just doesn't want to do video. |
| 23:48.44 | WIMPy | You can always play with interrupt smp affinity. |
| 23:48.48 | albania-asterisk | what do you mean ? |
| 23:49.25 | WIMPy | You can control which interrupts ar handled by which CPU(s). |
| 23:49.53 | volga629 | I have fanvil video phones here locally all working perfect voice and video the branch same phone which connect to asterisk through vpn tunnel don't want pass video properly |
| 23:50.03 | WIMPy | You could send all IRQs fromthe NIC to one core and make all others to others. |
| 23:50.42 | albania-asterisk | http://wikisend.com/download/325802/cat.txt |
| 23:50.45 | WIMPy | +go |
| 23:50.53 | volga629 | let me see if I can you tcpdump to catch SDP |
| 23:50.57 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 23:50.57 | volga629 | use |
| 23:51.12 | albania-asterisk | and why should i need to dedicate interrupts ? |
| 23:51.36 | WIMPy | It can halp under heavy load. |
| 23:51.38 | WIMPy | help |
| 23:52.00 | WIMPy | What was that link? |
| 23:52.20 | albania-asterisk | the result that i pasted but in notepad |
| 23:53.16 | WIMPy | I'm not going to accept cookies. |
| 23:53.45 | albania-asterisk | ok leave it is same thing as in paste bin , just as was formatted better |
| 23:54.29 | WIMPy | It was readable enough. |
| 23:54.43 | albania-asterisk | ok |
| 23:54.47 | WIMPy | But MSIs are never shared anyway. |
| 23:55.04 | ChannelZ | So according to RTP debug, I'm getting packets of 160 bytes for g711. Is that total or just the payload? |
| 23:55.26 | WIMPy | Ok, you never know what sick hardware someone comes up with. But it would be hard to do. |
| 23:55.44 | WIMPy | Musy be payload only. |
| 23:55.51 | albania-asterisk | are not shared for interrupts ? |
| 23:56.28 | WIMPy | It's something they finally got right with PCIe. |
| 23:56.29 | ChannelZ | heh yeah thankfully we're not still in the days of ISA |
| 23:56.52 | WIMPy | Took only 30 years to fix. |
| 23:57.39 | WIMPy | Well, in that respect ISA could be better than PCI. |
| 23:58.15 | ChannelZ | So payload only, 160 bytes of 8000/sec would be.. 50 packets a second? |
| 23:58.35 | WIMPy | Tht's the default. |
| 23:59.13 | albania-asterisk | so if i change server for cpu, what should check for this ? |
| 23:59.30 | ChannelZ | So for 1000 calls it's around 50k packets/sec just for one direction |