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| 01:27.20 | joecool | probably offtopic |
| 01:27.54 | joecool | but i have a cisco 3550 PWR 24 switch and it just seems to not want to power some polycom ip-501's |
| 01:28.11 | joecool | anyone familiar with this? |
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| 02:01.57 | ChannelZ | not specifically.. but I had some cheap Netgear desktop switches that were under-powered |
| 02:02.53 | ChannelZ | you couldn't use all the PoE ports at the same time as the load was too much. It was only rated for so much |
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| 02:17.04 | almostworking | join #ffmpeg |
| 02:17.07 | almostworking | ooops |
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| 02:22.05 | [TK]D-Fender | joecool, those only support Cisco's proprietary pre 802.3af PoE standard. You aren't going to get those phone running on that. |
| 02:22.17 | joecool | fml -_- |
| 02:22.28 | joecool | just picked up 5 of these polycoms |
| 02:22.46 | joecool | i have a cisco 7940 |
| 02:23.27 | joecool | well damn guess imma have to flip this worthless switch then and find one that handles the standard |
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| 03:27.00 | Synthase_ | joecool: You can change the inline power settings and likely get it to work. Have one on my desk that I've tested this with. |
| 03:28.19 | joecool | yeah i mean i'm reviewing polycom's literature |
| 03:28.36 | joecool | and they specifically say the phone is compatible with cisco pre-standard PoE |
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| 03:30.22 | Synthase_ | I have Polycom sound point something & Digium phones working on it, but all just bootlooped previously till changing interface power settings. |
| 03:30.36 | joecool | i don't even get a bootloop |
| 03:30.42 | joecool | it doesn't even show signs of powering |
| 03:30.48 | joecool | i've tried different cables, you name it |
| 03:31.16 | joecool | the only thing it even shows signs of powering is my swissvoice (that's dead), but the inline power light comes up |
| 03:31.34 | joecool | imma bring in a cisco 7940 tomorrow |
| 03:31.44 | joecool | and see if it powers that, if not i'm blaming the switch |
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| 03:59.21 | phix | CISCO is the worst |
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| 04:17.39 | Micc | Does asterisk 13 have a way to redirect current calls to another server? Say if I wanted to shut down the server but I didn't want to wait for calls to complete, could I pass those calls off to another server? Maybe with a script that did a reinvite on each channel or something? |
| 04:18.53 | Micc | I suppose that wouldn't really make sense, as there is application state that wouldn't transfer. |
| 04:21.21 | Micc | I guess it would be fine to leave those old calls on there until they were done. Like a core stop gracefully and redirect new calls to another server. |
| 04:21.59 | Micc | Ok, I think I've got it. Thanks for the help guys. ;) |
| 04:28.36 | joecool | i think i figured it out, i think the polycom needs a special pinout to work with cisco in-line power |
| 04:30.37 | joecool | yeah that's prob it |
| 04:30.44 | joecool | ugh, guess i'm making new cables tomorrow |
| 04:30.53 | joecool | better than buying new switch and/or phones though i suppose |
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| 04:59.03 | phix | Micc: Any time |
| 05:20.34 | [TK]D-Fender | Micc, "core show application transfer" |
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| 09:20.30 | slavon | Hello all |
| 09:22.36 | slavon | Please look to https://issues.asterisk.org/jira/browse/ASTERISK-24717 |
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| 10:35.09 | slavon | in topic "11.16.0". now 11.17.0 |
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| 11:59.56 | eschmidbauer | Hello |
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| 12:07.19 | racho | hello, i'm experiencing rtp clock drifts between two asterisk trunks...one is running 11.7.0 and the other 1.8.7.1. after resyncing with adjtimex and restarting the ntpd daemons on both machines i still get severe chopping in the beginning of the audio stream followed by delay from callee to calle. any ideas what can cause that aside from network status/connections? |
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| 12:13.08 | crocodilehunter | Hi Everyone. I have spent hours compiling, recompiling and searching Google for an answer, to no avail!! When I check the status of asterisk: #sudo service asterisk status; I get "asterisk dead but subsys locke"; It seems to be quite common but none of the solutions offered online seem to work.... Any help would be appreciated. I am running Asterisk 12.2.0 on a CentOS 6.5 (32 BIT).. |
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| 12:29.03 | avb | hey all |
| 12:29.06 | avb | from queue show with realtime configuration: member 1031^avb (SIP/1031 from SIP/1030) (ringinuse disabled) |
| 12:29.33 | avb | i had member interface set to SIP/1030 and after changed it to 1031 |
| 12:29.45 | avb | but queue still trying to dial 1030 |
| 12:31.02 | avb | queue reload is not fixing that, only asterisk restart |
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| 14:25.41 | phonesimon | good morning. i could use some help with the voicemail application if anyone's willing. VoiceMail() is recording messages left by G722 users in 8k wav format |
| 14:25.56 | phonesimon | this is OK if that's all it can do, |
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| 14:26.19 | phonesimon | but when the g722 users go to check voicemail from the device, asterisk won't transcode the 8k wav for them |
| 14:26.41 | phonesimon | ideally I would like the vm app to record in g722 and then transcode to lower quality if endpoints need it |
| 14:28.46 | phonesimon | what it does now when a g722 user goes into voicemailmain() is attempts to play back the file, looking for a g722 extension and then gives up when it does not find it |
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| 15:37.08 | Katty | PEANUT BUTTER CRACKERS. |
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| 15:45.48 | pjensen00 | No. That is not acceptable. |
| 15:49.35 | phonesimon | i'll offer PB crackers to whoever can answer the question i posed |
| 15:49.45 | phonesimon | that's cheap payment for (correct) advice |
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| 16:08.34 | [TK]D-Fender | phonesimon, it didn't record in g722 because you didn't tell it to in voicemail.conf |
| 16:08.57 | phonesimon | let me check |
| 16:09.01 | phonesimon | i did not see where that was set |
| 16:09.15 | phonesimon | only format= for wav, wav49, gsm |
| 16:09.23 | [TK]D-Fender | phonesimon, because it wasn't, otherwise you'd be getting g.722 versions as well |
| 16:10.12 | [TK]D-Fender | * will transcode from the best format available when playing them back. if native is available it will be chosen |
| 16:13.56 | phonesimon | OK, I could use a hint then. I don't see the option to specify recording formats. |
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| 16:15.04 | zekoZeko | phonesimon: you can try wav16 for format. |
| 16:15.18 | [TK]D-Fender | <phonesimon> only format= for wav, wav49, gsm <--- this IS IT |
| 16:15.40 | phonesimon | precisely... |
| 16:15.55 | [TK]D-Fender | no.. that IS the place to SPECIFY them |
| 16:16.00 | [TK]D-Fender | And you did NOT say g722 |
| 16:17.43 | phonesimon | Did I miss a previous answer to my question? yes, I wrote earlier that I want to record g722 format. |
| 16:17.52 | phonesimon | and want to play it back in the same format. |
| 16:18.00 | [TK]D-Fender | <[TK]D-Fender> <phonesimon> only format= for wav, wav49, gsm <--- this IS IT <----------------------- |
| 16:18.05 | [TK]D-Fender | Go. Set. It. There. |
| 16:19.21 | phonesimon | OK, i'll try it. Other information I saw said that those were the only options that could be specified (wav, wav49, gsm) |
| 16:19.55 | [TK]D-Fender | where do you see this? |
| 16:20.43 | [TK]D-Fender | Clearly not in the sample config which contradicts this premise you've been running on. |
| 16:21.11 | phonesimon | http://svnview.digium.com/svn/asterisk/trunk/configs/samples/voicemail.conf.sample?view=markup vs. http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf |
| 16:21.30 | [TK]D-Fender | Line 23 right there <- |
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| 16:22.43 | [TK]D-Fender | And that is a SAMPLE. It is not a list showing every possible combination, espcially for things like codecs where there are a dozen different possible values.. and then multiple that by how many different combinations you could make. |
| 16:23.28 | [TK]D-Fender | If thsoe were the only values that could be set then it also wouldn't even be an option. It'd just be a hard coded constant in the code and not in the config at all. |
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| 16:26.37 | phonesimon | OK, great. That works. Do you know why Asterisk was unwilling to transcode the 8khz wav file? |
| 16:27.24 | phonesimon | you mentioned "* will transcode from the best format available when playing them back. Â if native is available it will be chosen" â should not be a problem to transcode wav, right? |
| 16:27.27 | [TK]D-Fender | What did it play instead? |
| 16:28.49 | [TK]D-Fender | and it isn't a question of "problem". it is a question of transcode priority |
| 16:29.12 | phonesimon | Nothing. It skipped the message. Unable to open /var/spool/asterisk/voicemail/default/8274/Old/msg0000 (format (g722|h264)): No such file or directory |
| 16:29.34 | phonesimon | msg0000.wav is there |
| 16:30.04 | [TK]D-Fender | What version are you running? |
| 16:30.14 | phonesimon | 11 |
| 16:30.37 | [TK]D-Fender | You may not have the format module loaded. |
| 16:30.41 | [TK]D-Fender | go verify |
| 16:34.27 | phonesimon | format_wav, yes. |
| 16:35.02 | [TK]D-Fender | and codec formats for 722 |
| 16:35.48 | phonesimon | <PROTECTED> |
| 16:35.49 | phonesimon | for 722.1 |
| 16:37.52 | phonesimon | i do not see anything else appropriate â all are loaded |
| 16:37.59 | phonesimon | codec_g722 loaded also. |
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| 17:27.05 | CGMChris | I've got two seperate asterisk installations, both using the same SIP trunking provider. One works, the other does not. I'm trying to figure out why. On one installation, the provider seems to be appending a port number to the Call-ID, and I suspect that's the reason things aren't matching up. Does anyone know enough about how the packets are matched to confirm this? http://pastebin.com/1ajZ8phX |
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| 17:32.38 | CGMChris | And by doesnt work, I mean that inbound calls come into the PBX, hit the Answer() line, and then hangup as we are bombarded by the provider as it attempts to reroute the call through another of it's IPs. |
| 17:35.54 | CGMChris | As is illustrated here: http://pastebin.com/5znaTXz1 |
| 17:41.00 | CGMChris | Scratch that, bad example, I had peers commented out. The real example @ http://pastebin.com/ALRiJuhr |
| 17:48.47 | phonesimon | if something is changing the call-id, that something could be a SIP ALG on the router |
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| 18:00.34 | CGMChris | phonesimon: I've removed the router from the equation and my PBX is now plugged directly into the cable modem. I suspected ALG myself but that's not it. |
| 18:06.32 | phonesimon | can you connect to any other provider? |
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| 18:23.28 | CGMChris | phonesimon: Yes, we just ported to a new provider today. Everything was working fine until today, and we use the new provider on several other asterisk installs already with zero issues. |
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| 18:36.40 | mic_ | can you recommend a good call generator? something in the form of "StarTrinity SIP tester" - so that it is possible to simulate some weird loads and play different songs on certain connections. |
| 18:38.10 | robmal | sipp |
| 18:40.26 | mic_ | robmal: I was checking it - but last time I was looking at it, RTP was something "very new" ;( |
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| 19:50.00 | bobbymc | hi room, im trying to help this customer have better audio by tweaking the jiter settings for iax without effecting others on the same system |
| 19:50.12 | bobbymc | the are in mexico and have about 21ms avg jitter |
| 19:50.59 | WIMPy | Doesn't sound like much. But you shouldn't use jitter buffers in transit. They should be at the very end of the transmission and ONLY there. |
| 19:51.04 | bobbymc | anyone has any config suggestions? im not very comfortable tweaking them, there are 2 modes fixed and adaptive and a few others |
| 19:51.30 | bobbymc | they all have iax extentions |
| 19:52.07 | bobbymc | i know i can set the jitter buffer globally or on indevidual extentions right? |
| 19:52.13 | bobbymc | *extensions |
| 19:52.30 | WIMPy | The only situation where a JB on IAX makes sense if you receive calls via IAX and pass them on to TDM hardware. |
| 19:53.27 | bobbymc | i see, they are limited in bandwidth, i currently use ilbc for the codec |
| 19:53.38 | bobbymc | a lot of their calls sound like robots |
| 19:54.02 | WIMPy | Maybe that's more of a packet loss issue? |
| 19:54.30 | bobbymc | i think so, but they cant upgrade the internet anymore then they already have, just trying to find any other way to help |
| 19:55.52 | WIMPy | The issue with packet loss due to low badwidth is as follows: You can increase packetization to save a little badwidth, making PL less likely. The catch is that the impact of a lost packet will be |
| 19:55.55 | WIMPy | higher. |
| 19:56.18 | WIMPy | Do you have trunk mode enabled? |
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| 23:43.42 | EOIP | Hi Everyone. Is this 'Asterisk is dead but subsys locked' a problem? It comes and goes. The only evident problem is it sometimes says that it fails to shut down but the Asterisk console seems to work. I suspect it's from SSHing to the Asterisk box from different machines... (Asterisk 12.2.0 on 32 bit centOS 6.5) |
| 23:44.43 | EOIP | Sorry, 'Asterisk is dead but subsys locked' gets printed when I do #sudo service asterisk status |
| 23:57.21 | robmal | Poor asterisk. Did you check /var/log/asterisk/messages for the death certificate? |
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