| 00:01.45 | *** join/#asterisk bhans (ddfd4592@gateway/web/freenode/ip.221.253.69.146) |
| 00:04.41 | *** join/#asterisk acidfu_ (~acidfoo@24-212-247-227.cable.teksavvy.com) |
| 00:36.12 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
| 00:38.08 | WIMPy | Hmm. Is the only way to send a registration to a non-standard port, to configure outboundproxy? Doesn't port do anything? |
| 01:01.22 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
| 01:11.53 | *** join/#asterisk acidfu_ (~acidfoo@24-212-247-227.cable.teksavvy.com) |
| 01:27.51 | Penguin | You want to register your asterisk to some remote system on a different port? |
| 01:28.28 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
| 01:29.30 | WIMPy | yes |
| 01:32.27 | Penguin | In the peer entry for that peer, set port=somePort |
| 01:32.49 | WIMPy | Doesn't seem to do anything. |
| 01:33.03 | WIMPy | It still sends to 5060. |
| 01:33.45 | Penguin | The register statement may also need the new port. |
| 01:34.08 | Penguin | register => user:secret@host.domain.com:port |
| 01:34.13 | WIMPy | hasn't used register statements for ages. |
| 01:34.37 | Penguin | If you're sending SIP registrations out to them, you'll probably need to set the port on that line as well. |
| 01:35.11 | Penguin | I have an ITSP that has non-standard ports available for bypassing ISP's firewalls, but I've never had to use the non-standard ports. |
| 01:35.22 | WIMPy | I meant that it doesn't exist. I've been using callbackextension since I found it. |
| 01:36.57 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 01:37.35 | Penguin | ; A similar effect can be achieved by adding a "callbackextension" option in a peer section. |
| 01:37.38 | Penguin | ; (note that the "port" is ignored - this is a bug that should be fixed). |
| 01:38.10 | Penguin | Until the bug is fixed, I guess you'll have to resort back to the old register statement with non-standard port. |
| 01:38.10 | WIMPy | Guess that never happened. |
| 01:38.47 | UncleKiwi | hi - I seem to getting a bit of this '105 No Authentication ' when i go 'sip show registry' not sure whats going on |
| 01:38.54 | WIMPy | outboundproxy also works. Both feel a little iffy, but well... |
| 01:39.14 | Penguin | If it will work reliably, I say go for it. |
| 01:41.28 | Penguin | I can't use callbackextension because none of my register statements define an extension after the hostname to register against. |
| 01:41.37 | Penguin | Unless I don't understand how it works. |
| 01:51.46 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 01:54.27 | UncleKiwi | Asterisk 11.22.0 built by root @ rpi2 on a armv7l running Linux on 2016-07-17 03:28:53 UTC |
| 01:55.13 | UncleKiwi | seems to lose registration and then show the above error and does not regain registration until i reload sip |
| 02:07.06 | *** join/#asterisk Oatmeal (~Suzeanne@c-68-51-46-116.hsd1.in.comcast.net) |
| 02:37.10 | UncleKiwi | opps its this one Asterisk certified/11.6-cert13 |
| 02:49.23 | drmessano | Why are you using Certified? |
| 02:49.40 | drmessano | Use current unless you have a support need |
| 02:50.33 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 02:53.00 | UncleKiwi | ok |
| 02:53.18 | UncleKiwi | i guess i was unsure |
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| 03:44.46 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 04:00.37 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:40.02 | *** join/#asterisk FarhaadN (~Farhad@82.99.206.194) |
| 04:44.22 | FarhaadN | hi everyone,does anyone know why my sip channesl dont hangup, i use asterisk 11.23.0 and voice gateway cisco |
| 04:44.33 | FarhaadN | this is a asterisk bug or cisco |
| 04:44.34 | FarhaadN | ? |
| 04:46.15 | voipmonk | please show the evidence |
| 04:47.13 | FarhaadN | what u want? |
| 04:47.20 | FarhaadN | core show channels? |
| 04:48.41 | FarhaadN | this problem happend for many time, and every time i show hangup the channel in asterisk cli |
| 04:50.54 | FarhaadN | voipmonk: i still have a old problem with InUse |
| 04:51.35 | FarhaadN | in that case, probel is rather than this,and there are't any channel |
| 04:51.54 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-untebqgtyuvcfund) |
| 05:22.26 | [TK]D-Fender | voipmonk> please show the evidence <--- |
| 05:24.04 | wyoung | FarhaadN: sip set debug on |
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| 07:00.46 | *** join/#asterisk war9407 (war@static-72-73-18-14.clppva.fios.verizon.net) |
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| 07:17.24 | tirej | hi everyone |
| 07:18.08 | tirej | has anyone used AsterNET before? |
| 07:18.26 | tirej | can we build a complex-complete interface with that? |
| 07:20.04 | *** join/#asterisk jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za) |
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| 07:49.17 | *** join/#asterisk bounceman (~bounceman@185.32.9.250) |
| 07:50.44 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 07:51.48 | bounceman | Hi, I am wondering what causes Asterisk to return SIP 488 on in-dialog INVITE? And what is the consequence of this behavior? Thank you very much for any replies |
| 07:54.46 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:56.57 | bounceman | We are having these issues with FAX |
| 07:57.00 | bounceman | I found this in the CLI |
| 07:57.05 | bounceman | [Aug 24 09:34:17] WARNING[6382][C-000acfbd] chan_sip.c: Failed to initialize UDPTL, declining image stream |
| 07:57.05 | bounceman | [Aug 24 09:34:17] WARNING[6382][C-000acfbd] chan_sip.c: Failing due to no acceptable offer found |
| 07:59.17 | HRH_H_Cr1b | hello all |
| 07:59.27 | HRH_H_Cr1b | http://paste.debian.net/791251/ |
| 08:00.05 | HRH_H_Cr1b | am i right in thinking that if the call is successful, at line 5, we will never actually get to ok2 (line 6) as one or other party will hang up thus ending the dialplan at that point? |
| 08:00.20 | HRH_H_Cr1b | (i would need to use the 'h' extension to continue processing i guess?) |
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| 08:21.22 | *** join/#asterisk apb1963 (~apb@107-146-220-94.res.bhn.net) |
| 08:41.50 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
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| 09:54.40 | FarhaadN | hi, i call cell phone that is off , but ext still ringing on call |
| 09:54.42 | FarhaadN | why? |
| 09:58.40 | stefan27 | dialplan matching does not support matching 1 or more of preceeding character? |
| 09:59.41 | stefan27 | like _3¤4 matching 34 334 3333334 but not 4 354 |
| 10:00.22 | stefan27 | like * and + is to standard (sed-like) regexps? |
| 10:04.40 | *** join/#asterisk ruied (~ruied@125.44.62.94.rev.vodafone.pt) |
| 10:32.25 | TandyUK | stefan27: _[3].4 perhaps? |
| 10:32.36 | TandyUK | ah ignore that |
| 10:44.28 | Samot | FarhaadN: Perhaps the carrier is sending back a ringing while it checks to see if the cell phone is "on" |
| 10:44.33 | Samot | Or accepting calls. |
| 10:47.00 | Samot | bounceman: You need t38pt_udptl set. |
| 11:07.06 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 11:08.11 | stefan27 | Whenever a local channel Loc_X performs Dial(SIP/A) is it the channel variable "Extension" of Loc_X that determines the value of the channel variable "callerid" on the created channel SIP/A-0000003 (unless peer A has static callerid configured in sip.conf)? |
| 11:10.30 | stefan27 | or is it Loc_X's "Connected Line ID"? |
| 11:41.11 | FarhaadN | Samot: carrier send me that is busy,beacuse when i call cell phone with another number and out of my call center,this was ok, any configure for cisco not need? |
| 11:42.20 | bounceman | Samot: thank you. I am wondering. We use Asterisk as a B2BUA and when we recieve incoming fax they have all the good looking T38 SDP. But then other leg's SDP look just like a simple call. Is that an issue? |
| 11:47.13 | *** join/#asterisk acidfu_ (~acidfoo@modemcable002.114-70-69.static.videotron.ca) |
| 11:56.04 | Samot | bounceman: You mean the call from the carrier has all the T.38 media in it? |
| 11:56.14 | Samot | What is the "other leg"? |
| 11:57.43 | bounceman | SRC <--LEG1--> * <--LEG2--> DST |
| 11:57.58 | bounceman | First leg has T38 info in the SDP, LEG2 does not |
| 12:01.14 | bounceman | Maybe it is not needed and asterisk does all the translation, I have no clue. |
| 12:01.17 | bounceman | I do not like FAX |
| 12:09.43 | Samot | No, you need to pass it on to the endpoint with t38 |
| 12:13.39 | bounceman | What happens if I dont? |
| 12:14.27 | Samot | Then no fax. |
| 12:14.41 | Samot | You have the issue that you have now. |
| 12:14.49 | Samot | Your PEER with your provider needs t38 |
| 12:15.01 | Samot | Any endpoint should have it enabled as well. |
| 12:15.33 | bounceman | Well I have solved my first issue with the errors using t38pt_udptl. But the issue with SDP is still different between the legs, would you say this is a issue? |
| 12:15.47 | Samot | Where are the faxes going? |
| 12:15.51 | Samot | To an ATA? |
| 12:15.55 | *** join/#asterisk bmg505 (~leon@165.255.123.223) |
| 12:16.57 | bounceman | Something analog, that is all I know |
| 12:17.32 | Samot | Well it needs an adapter. |
| 12:17.39 | Samot | Like a SPA112 or something. |
| 12:17.49 | Samot | So that should be setup to accept t38 as well. |
| 12:18.24 | bounceman | Well, it is a setup that are accepting faxes without any issues. However when it pass through our B2BUA it works VERY sporradic. |
| 12:18.40 | Samot | Well there's something about FAX that has never changed. |
| 12:18.44 | Samot | FAX is best effort. |
| 12:18.48 | Samot | Always has been. |
| 12:18.51 | Samot | Even on POTS. |
| 12:19.26 | Samot | I would need to see the SIP debug of a FAX that failed. |
| 12:19.49 | bounceman | I can provide that |
| 12:20.16 | bounceman | Would you like a pcap or txt? |
| 12:33.25 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 13:01.53 | stefan27 | if there's a sip peer A on IP X and a sip friend B that has registered from IP X and asterisk receives an invite from IP X, which of A and B does asterisk try to match first? |
| 13:02.09 | stefan27 | I can't remember and I forgot where to look up this information other than digging into source code |
| 13:03.25 | [TK]D-Fender | The physically first one loaded from the config |
| 13:03.52 | [TK]D-Fender | and that will eb the only thing it amtches against. It'll never hit the 2nd |
| 13:04.17 | [TK]D-Fender | Which is why when you have multiple devices behind the same IP you should be using FRIEND, not PEER. |
| 13:04.27 | [TK]D-Fender | since that matches first on the user |
| 13:15.10 | *** join/#asterisk matrix1233 (~matrix123@197.3.206.61) |
| 13:26.11 | *** join/#asterisk afournier (~admin@80.215.212.130) |
| 13:26.16 | afournier | hi |
| 13:27.34 | afournier | does someone know about bridge that would "mix" different channels' audio based on VAD ? i.e to handle ConfBridge for G729 with zero transcoding |
| 13:28.21 | stefan27 | thx |
| 13:28.42 | afournier | it would do the samething softmix does for video but for audio (more a selecter than a mixer) |
| 13:28.42 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-ilrgytghakealrjm) |
| 13:28.43 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 13:42.03 | *** join/#asterisk RahaiL (~RahaiL@73.161.239.35) |
| 13:42.34 | RahaiL | Hi there how are you I got wired quesiton.. We are using Asterisk with one of the CDMA network as soon as call HIT CDMA network it give answer |
| 13:42.50 | RahaiL | and asterisk start give answer signal to other switch |
| 13:42.59 | RahaiL | is there way in asterisk we can give delay giving out answer |
| 13:43.04 | RahaiL | signal to other server |
| 13:43.26 | [TK]D-Fender | You haven't shown us the call so we can see what's happening or even told us how * is ttalking to it. |
| 13:44.10 | RahaiL | Asterisk box ==> CDMA GATEWAY |
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| 13:46.53 | *** join/#asterisk SunGod (~SunGod@92.55.116.125) |
| 13:49.37 | [TK]D-Fender | Try again.... |
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| 13:59.42 | *** join/#asterisk ruied (~ruied@125.44.62.94.rev.vodafone.pt) |
| 14:06.53 | RahaiL | not sure how to expaling just looking for way to send delay on answer signal |
| 14:08.20 | [TK]D-Fender | [TK]D-Fender> You haven't shown us the call so we can see what's happening or even told us how * is ttalking to it. |
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| 16:29.59 | *** join/#asterisk chl_ (~user@unaffiliated/chl/x-9330839) |
| 16:30.43 | chl_ | hello |
| 16:31.07 | chl_ | I have a device which is registrered, but asterisk log keeps telling me it failed to register |
| 16:31.27 | chl_ | .. with the error "wrong password" |
| 16:34.23 | *** join/#asterisk lotsofcows (~lotsofcow@host86-149-6-190.range86-149.btcentralplus.com) |
| 16:36.14 | lotsofcows | We had an internet outage. Once connectivity was back, we could make outgoing calls. Incoming calls died with "Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)". Had to restart asterisk. What should I have done? |
| 16:37.38 | *** join/#asterisk matrix1233 (~matrix123@197.3.206.61) |
| 16:40.49 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 16:44.43 | chl_ | i fixed it |
| 16:50.42 | [TK]D-Fender | lotsofcows, restarting should have no impact on that. 20 = no IP to contact them at. So whatever you were dialing hadn't registered, had timed outm, or other issue |
| 16:50.51 | [TK]D-Fender | * has nowhere to call. |
| 16:50.53 | [TK]D-Fender | That sums it up |
| 16:55.59 | wyoung | hey Mr [TK]D-Fender |
| 16:56.01 | *** join/#asterisk lotsofcows (~lotsofcow@host86-159-249-234.range86-159.btcentralplus.com) |
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| 17:01.37 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 17:01.37 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.10.0 (2016/07/21), 11.23.0 (2016/07/21), Standard: 14.0.0-beta1 (2016/07/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 17:06.42 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 17:09.01 | *** join/#asterisk zway (~zway@74.203.105.194) |
| 17:12.47 | ruied | Hello. I'm having a strange problem. I have two trunks Account_A and Account_B to make/receive calls fom the same provider (with different numbers). when I receive the call fom Account_B asterisk always wants to use the context defined in Account_A (context_from-pstn) but it should use the context from-CAIP01. http://pastebin.com/kJSuScNT |
| 17:13.56 | ruied | do not know if I'm missing something or if it is some kind of bug... |
| 17:14.22 | [TK]D-Fender | * will match the FIRST peer with a matching host= and that is all |
| 17:14.43 | [TK]D-Fender | Since they are both the same ... automatic fail |
| 17:14.52 | [TK]D-Fender | that will not be a way to separate the calls |
| 17:17.13 | *** join/#asterisk infobot (ibot@rikers.org) |
| 17:17.13 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.10.0 (2016/07/21), 11.23.0 (2016/07/21), Standard: 14.0.0-beta1 (2016/07/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 17:18.03 | *** join/#asterisk infobot (ibot@rikers.org) |
| 17:18.03 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.10.0 (2016/07/21), 11.23.0 (2016/07/21), Standard: 14.0.0-beta1 (2016/07/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 17:24.40 | ruied | how can I differentiate them ? |
| 17:25.41 | *** join/#asterisk infobot (ibot@rikers.org) |
| 17:25.41 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.10.0 (2016/07/21), 11.23.0 (2016/07/21), Standard: 14.0.0-beta1 (2016/07/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 17:26.43 | [TK]D-Fender | look at the call once it has arrived |
| 17:26.53 | [TK]D-Fender | the EXTEN should be your first hint |
| 17:34.27 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-15.wall.uls.co.za) |
| 17:38.36 | lotsofcows | Incoming calls dying with "Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)". Any ideas? |
| 17:39.35 | [TK]D-Fender | that has nothing to do with incoming calls |
| 17:39.44 | [TK]D-Fender | that is a message for a call you are trying to dial OUT |
| 17:39.54 | [TK]D-Fender | Go look at the call and then at what you are dialing out |
| 17:48.42 | *** join/#asterisk piling (~piling@unaffiliated/piling) |
| 18:07.14 | *** join/#asterisk SunGod (~SunGod@92.55.116.125) |
| 18:07.25 | ruied | [TK]D-Fender, so, I can't set each account to different contexts? |
| 18:07.52 | [TK]D-Fender | There is no identifying them as being separate that way |
| 18:07.57 | [TK]D-Fender | I thought I was very clear on this |
| 18:08.22 | [TK]D-Fender | it will take the FIRST one whose host matches and that's it |
| 18:08.35 | [TK]D-Fender | You then need to parse out what to do based on the content of the invite |
| 18:09.03 | [TK]D-Fender | <[TK]D-Fender> look at the call once it has arrived |
| 18:09.03 | [TK]D-Fender | <[TK]D-Fender> the EXTEN should be your first hint |
| 18:09.04 | [TK]D-Fender | ^^^ |
| 18:31.10 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-bxqjtmfcmtzfswdd) |
| 18:31.19 | *** join/#asterisk apb1963_ (~apb@107-146-220-94.res.bhn.net) |
| 18:35.46 | ruied | I know how I can deal with the incoming numbers if both trunks goes to "from-pstn" context. I wanted to separate them to different contexts (from the sip.conf), didn't know if you were pointing me to deal with the incoming number (in the same context at extensions.conf from-pstn context) or if you were trying to tell me that there is some way to set them to different contexts. English is not my main language... |
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| 20:29.19 | garygraham | Hey all, I have a system that makes outgoing calls to users, and plays an mp3 when they pick up. However I'm running into issues with voicemail. If it goes to voicemail, it seems that the message starts playing immediately, and doesn't wait for the beep. Is there some canonical way of dealing with this? |
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| 20:30.44 | [TK]D-Fender | * will never know to wait for a beep |
| 20:30.52 | [TK]D-Fender | Thre is no sane way to do this |
| 20:31.24 | [TK]D-Fender | The most you can do is use AMD and then a WAITFORSILENCE to try to guess when to actually start |
| 20:33.06 | garygraham | Thanks for the quick response! Just wondering how telemarketers manage to do it so effectively with robocalls. I suppose they aren't using asterisk :P I will dig into it, thanks again. |
| 20:33.55 | [TK]D-Fender | You're welcome |
| 20:34.46 | [TK]D-Fender | This method is about the only way I can think of given even having a beep is optional, with varying potential frequencies, etc |
| 20:34.57 | [TK]D-Fender | I'd bet on the same methodology |
| 20:35.10 | [TK]D-Fender | heads home |
| 20:39.10 | Kunsi | garygraham: my experience with telemarketers and voicemail is exactly what you are experiencing: if you listen to your voicemail, it just starts in the middle of the prerecorded text |
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| 20:58.19 | matt_ | does anybody know if its possiable to get asterisk to return 484 address incomplete for a user? |
| 20:59.33 | matt_ | or do I have to match anything shorter than a valid ext and use something like ,Incomplete() ? |
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| 22:25.32 | Mango45 | I'm having a problem with choppy recordings. One side of the call is over the internet, so it is possible there is a bandwidth issue there. The other side of the call is on the LAN. There's plenty of LAN bandwidth available, and the server has plenty of resources, yet the LAN side of the call is also choppy. I cannot reproduce the problem and have no idea when it will occur. |
| 22:26.31 | robmal | tcpdump and wireshark |
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| 22:28.16 | Mango45 | I also have the problem when the internet user is speaking to our IVR, so it's not even over the LAN. |
| 22:32.37 | Mango45 | I made a test call and manufactured some packet loss on one side. The other side of the recording was fine, so that is not the problem. |
| 22:32.45 | Mango45 | Could it be a timing issue and if so how do I troubleshoot that? |
| 22:34.17 | robmal | What kind of connection do you have? |
| 22:34.41 | Mango45 | The internet portion of the call is ADSL. |
| 22:35.03 | Mango45 | The PSTN portion of the call is via a VoIP service provider with SIP. |
| 22:37.19 | robmal | Any QoS on this ADSL? |
| 22:37.52 | Mango45 | Yes. However, I'm more concerned about choppy audio on the recording of the IVR-side of the call. |
| 22:38.27 | Mango45 | Asterisk is using Monitor to record the call and Playback to play some audio, and the recording is choppy. |
| 22:40.46 | Mango45 | I don't have any timing modules loaded; do I need that? |
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| 22:51.21 | Mango45 | The only functionality that requires internal timing is IAX2 trunking. It may also be used when generating audio for playback, such as from a file. Even though internal timing is not a requirement for most Asterisk functionality, it may be advantageous to use it since the alternative is to use timing based on incoming frames of audio. If there are no incoming frames or if the incoming frames of audio are from an unreliable or jittery source, then the corr |
| 22:51.25 | Mango45 | Sounds reasonable; hope this helps. |
| 22:56.00 | matt_ | Mango45: what ext have you got for the filename of the MixMonitor statement? |
| 22:59.23 | matt_ | Mango45: ive had funny issues when i had an ext that needed to be transcoded from the endpoints, these days I force everything to alaw and give a filename ending in .alaw |
| 22:59.41 | matt_ | to avoid transcoding, dont have any issues anymore |
| 23:06.50 | Mango45 | Good point. It's in wav. |
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| 23:12.24 | matt_ | Mango45: ok, will probuly need to transcode then |
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| 23:13.18 | matt_ | Mango45: also, incase it helps when i save to alaw to play it back i use aplay -f a_law in.alaw and to re encode it to wav i use ffmpeg -ar 8k -f alaw -i in.alaw out.wav |
| 23:13.31 | matt_ | anyway, i need to sleep now |
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| 23:14.28 | Mango45 | Thanks. I will give that a shot. |
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