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| 01:49.49 | drmessano | hdon: Nope |
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| 02:56.16 | volga629 | Hello Everyone, trying configure on asterisk 14 PUBLISH for bria soft client and getting error 489 with message res_pjsip_pubsub.c:3183 pubsub_on_rx_publish_request: No registered publish handler for event presence |
| 02:56.20 | CRCinAU | ok, so I have a SIP register question.... I set defaultexpiry=600 in sip.conf, then register with two different SIP providers... |
| 02:56.53 | CRCinAU | I note that in 'sip show registry' that one provider stays at a refresh of 105, the other at 585. |
| 02:57.18 | Samot | That's how long is left. |
| 02:57.22 | CRCinAU | would I be correct in assuming that the REGISTER message contains an expiry, which can be either accepted or overriden by the remote end? |
| 02:57.59 | volga629 | that the config https://paste.fedoraproject.org/paste/C0TeeLh-o5t4cA519cvh2g |
| 02:58.24 | CRCinAU | so also assuming that one VSP uses 120 seconds ttl, whereas the other will allow whatever I set (up to some unknown threshold)? |
| 03:01.31 | *** join/#asterisk LiuYan (~NiHola@unaffiliated/liuyan) |
| 03:02.16 | Samot | What is the issue you are having with the REGISTERs? |
| 03:02.32 | CRCinAU | I'm just trying to complete my knowledge. |
| 03:03.08 | CRCinAU | I've looked at the Expires in the RFC: https://tools.ietf.org/html/rfc3261#section-20.19 |
| 03:03.16 | Samot | The side that is REGISTERing has to send the Expire time |
| 03:03.30 | CRCinAU | yet that isn't informative for implementation / usage |
| 03:04.14 | CRCinAU | and the other side can either accept that, or set a new Expires for the registration timeout? |
| 03:04.35 | Samot | The other side generally just accepts it |
| 03:04.51 | Samot | Otherwise they can't guarantee they're sending requests to the right location all the time. |
| 03:05.54 | CRCinAU | See, the RFC also says: |
| 03:05.56 | CRCinAU | 10.2.1.1 Setting the Expiration Interval of Contact Addresses |
| 03:05.58 | CRCinAU | When a client sends a REGISTER request, it MAY suggest an expiration interval that indicates how long the client would like the registration to be valid. (As described in Section 10.3, the registrar selects the actual time interval based on its local policy.) |
| 03:06.55 | LiuYan | Hi, is it possible to trigger a sub-dialplan when call alerting event received (Dial)? |
| 03:07.28 | Samot | Correct. |
| 03:07.48 | Samot | LiuYan: You can trigger a Maco and or GoSub |
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| 04:36.07 | LiuYan | Samot: Thanks, I just read the help of M() option of Dial application, it doesn't said it's triggered by Alerting event, or is it? |
| 04:36.57 | LiuYan | need Alerting event |
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| 04:39.05 | volga629_ | Hello Everyone, how to create outbound publish to pull device state from kamailio ? |
| 04:41.10 | Samot | How is that an Asterisk issue? |
| 04:41.30 | volga629_ | yes I don't see it send PUBLISH to kamailio |
| 04:41.42 | Samot | You said from |
| 04:41.44 | Samot | Not to |
| 04:41.47 | Samot | Which is it? |
| 04:41.57 | Samot | You want Asterisk to send a PUBLISH to Kamailio? |
| 04:45.04 | volga629_ | yes kamailio handle all presence asterisk should be able pull user state through outbound-publish |
| 04:45.35 | volga629_ | this is from wiki |
| 04:45.36 | volga629_ | This module works perfectly with the PJSIP extension state publishing support. The Asterisk configuration needs to use a URI to the Kamailio server and the Kamailio server has to explicitly trust traffic from the Asterisk instance, or authentication needs to be configured. |
| 04:47.23 | Samot | Your context is junk. |
| 04:47.31 | Samot | It's not even complete. |
| 04:48.52 | CRCinAU | I'm wondering... is there a pjsip equiv of 'insecure=INVITE'? |
| 04:49.46 | CRCinAU | before I went back to chan_sip, my issue with pjsip was that if I have two peers that I register with, it was asking for auth on an INVITE for a call - but seemingly matching it against the wrong definition on my end - therefore failing the auth... |
| 04:50.09 | CRCinAU | ie I have 2 accounts I REGISTER to - only differing in username & password. |
| 04:51.50 | CRCinAU | I'd have to dig out my config file to answer specifics... but yeah... |
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| 04:54.58 | CRCinAU | jcolp (whoever taht is) seems to say: We'd need to see the actual full configuration for everything and the logs to go deeper into this. I will say for the match - you can't have two match with the same IP address going to different endpoints. It won't work. |
| 04:55.12 | Samot | https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample |
| 04:56.10 | CRCinAU | I think my point is that it doesn't seem like you can tell the difference between two peers on the same IP within pjsip |
| 04:57.55 | CRCinAU | bbl... meetings :\ |
| 05:04.59 | drmessano | Uh |
| 05:05.05 | drmessano | You couldnt with chan_SIP either |
| 05:05.37 | drmessano | They have the same IP |
| 05:05.49 | Samot | This is 100% a configuration issue. |
| 05:30.37 | [TK]D-Fender | You can ..... FRIEND.... is your friend |
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| 07:16.27 | CRCinAU | yay. meeting over. |
| 07:16.36 | CRCinAU | so yeah - insecure=INVITE causes chan_sip to work |
| 07:16.48 | CRCinAU | but there doesn't seem to be an equiv in pjsip at all :\ |
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| 07:17.17 | CRCinAU | it asks for auth for the INVITE, which is ok - but the auth fails - so I'm guessing its attempting the auth for the other account. |
| 07:17.31 | CRCinAU | as it all works if you only register with ONE service on ONE IP |
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| 07:24.56 | Hydrosine | morning! I have an asterisk 13.17.2 with realtime pjsip.. now one strange thing happens. when I set a fromdomain on an endpoint in the DB. the resulting dial on that endpoint has brackets in the from. such as: From: <sip:137@[pbx1.testdomain.com:6106]>;tag=1b5a8396-eec0-47c7-8261-f75e3d05dae7 |
| 07:25.34 | Hydrosine | obviously upstream does no accept this formatting :) i tried switching between the ansi and utf module for odbc, but makes no difference.. |
| 07:32.09 | Hydrosine | removing the port from , fromdomain helps ;) |
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| 08:57.47 | TheGallopingFox | is there a way to show a name instead of a extension on callerid? |
| 08:58.45 | TheGallopingFox | ie. 1001 shows Joe Blogg when it dials another phone |
| 08:59.20 | TheGallopingFox | i guess it would be a setting in sip.conf ? |
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| 09:11.09 | sekil | hello |
| 09:17.17 | LiuYan | TheGallopingFox: Set(CALLERID(name)=THE_NAME_YOU_WANTED_TO_SHOW) do the trick |
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| 09:32.18 | TheGallopingFox | thanks |
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| 14:43.09 | Samot | Oh file: I can do Dial(SIP/{peer}/{exten}!@{todomain}!{fromuser}@{fromdomain})? {} = placeholder in this format. |
| 14:43.20 | Samot | I don't need to include a touser or do I? |
| 14:43.23 | file | I don't know chan_sip |
| 14:43.29 | file | not that part of it at least |
| 14:43.32 | Samot | You suck. |
| 14:43.43 | Samot | OK, I'll give it a go. |
| 14:44.07 | Samot | The way it breaks down in the sample config...I think it's doable. |
| 14:44.40 | Samot | [![touser[@todomain]][![fromuser][@fromdomain]]] <-- the way the brackets are gives me the indication they can be separate |
| 14:47.33 | TheGallopingFox | my setup works fine, however when i try and call a smartphone through openvpn, it registers and rings, but there is no audio? |
| 14:47.42 | TheGallopingFox | *softphone |
| 14:50.24 | TheGallopingFox | tcpdump shows rtp packets coming from openvpn ip |
| 14:50.34 | TheGallopingFox | but im not sure why there isnt any audio? |
| 14:53.32 | [TK]D-Fender | "your setup is fine " doesn't really mean anything when you then tell us your call has no audio. |
| 14:53.41 | [TK]D-Fender | We should be looking at actual full call debug. |
| 15:01.13 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 15:05.41 | TheGallopingFox | fixed it! directmedia=no |
| 15:16.19 | TheGallopingFox | are there any downsides to using directmedia=no ? |
| 15:16.40 | Samot | Outside of it being the preferred method, no. |
| 15:16.57 | TheGallopingFox | good :) |
| 15:17.25 | TheGallopingFox | so thats it then, a fully functional asterisk voip setup |
| 15:17.28 | [TK]D-Fender | "downsides"..... |
| 15:17.50 | [TK]D-Fender | I guess the downside of not doing it means your audio isn't flowing.... |
| 15:17.52 | Samot | Direct media means endpoint to endpoint |
| 15:17.59 | Samot | If they are both behind NAT, it sucks |
| 15:18.16 | Samot | If they have different codec choices that don't match up, no call |
| 15:18.35 | *** join/#asterisk voipmonk (~voipmonk@138.68.2.1) |
| 15:18.44 | Samot | The point being a B2BUA, like Asterisk, is to avoid those issues. |
| 15:18.55 | [TK]D-Fender | So letting them go direct risks them failing. NOt going direct forces media through your server ... which if they CAN support it does add more traffic to your server, and a little CPU |
| 15:19.45 | TheGallopingFox | oh ok, so its better for directmedia to be off |
| 15:19.56 | Samot | Like I said. |
| 15:20.02 | Samot | It is the *preferred* method. |
| 15:20.03 | [TK]D-Fender | You want audio right? |
| 15:20.11 | TheGallopingFox | yah |
| 15:20.30 | Samot | directmedia=no or direct_media=no <-- depending on your driver. |
| 15:20.33 | [TK]D-Fender | Letting your devices think they handle their own business tends to turn out being false |
| 15:21.24 | TheGallopingFox | can directmedia=no effect call quality? |
| 15:21.46 | TheGallopingFox | maybe little bit of latency? |
| 15:22.29 | [TK]D-Fender | yes, obviousl a possibility of latency. |
| 15:22.48 | [TK]D-Fender | if both ends are across the internet. |
| 15:23.07 | Samot | Two endpoints behind NAT talking to each other directly = BAD NEWS |
| 15:23.32 | Samot | It is literally the one thing to avoid. |
| 15:23.43 | TheGallopingFox | i noticed the HD icon doesnt show now on polycom 335 when i make a call :-/ |
| 15:23.55 | TheGallopingFox | ok |
| 15:24.02 | Samot | Do you have G722 enabled on Asterisk? |
| 15:24.06 | TheGallopingFox | yes |
| 15:24.10 | Samot | It is part of the allowed codecs? |
| 15:24.13 | TheGallopingFox | yes |
| 15:24.29 | Samot | What codec is being used for the call now? |
| 15:24.41 | TheGallopingFox | im not sure how to find that out |
| 15:24.48 | Samot | sip set debug on |
| 15:24.52 | Samot | Look at the SDP |
| 15:25.32 | [TK]D-Fender | There is "allowed", and there is "PREFERRED". |
| 15:25.38 | [TK]D-Fender | Order matters more than a little |
| 15:25.41 | [TK]D-Fender | on both sides |
| 15:26.15 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148) |
| 15:28.14 | TheGallopingFox | lots of info went down the screen but i think i saw G.722 |
| 15:28.35 | Samot | ~pb |
| 15:28.35 | infobot | [pastebin] 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:28.36 | Samot | show it |
| 15:28.42 | TheGallopingFox | a=rtpmap:9 G722/8000 |
| 15:28.49 | Samot | No. |
| 15:28.50 | Samot | Show it all |
| 15:28.53 | Samot | So we can see the order |
| 15:28.57 | Samot | And offer/answer |
| 15:29.02 | TheGallopingFox | ok how can i paste it |
| 15:29.12 | Samot | Seriously? |
| 15:29.17 | Samot | ~pb |
| 15:29.18 | infobot | somebody said pastebin was 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:29.30 | TheGallopingFox | i need to pump it into a txt file first |
| 15:29.40 | Samot | Scroll back |
| 15:29.42 | Samot | Copy |
| 15:29.44 | Samot | Paste |
| 15:30.01 | TheGallopingFox | not working in this screen session |
| 15:30.24 | [TK]D-Fender | Fuck "screen" |
| 15:30.29 | TheGallopingFox | :o |
| 15:30.31 | [TK]D-Fender | Seriously |
| 15:30.51 | [TK]D-Fender | I mean that whole-heartedly |
| 15:31.13 | [TK]D-Fender | Get real terminal output you can capture |
| 15:31.27 | [TK]D-Fender | putty makes this a 2-click job |
| 15:32.38 | TheGallopingFox | ok will change terminal |
| 15:37.01 | Samot | file: Answer is, you need both. It works, just sends the TO user as empty though. |
| 16:02.00 | *** join/#asterisk DanB (~DanB@clt-195.192.207.35.ip-anschluss.net) |
| 16:09.32 | Samot | I love when end users decide giving me the whole story behind "I want to blacklist these numbers" is the best way to get the request done. |
| 16:10.30 | Samot | When the story started to trigger thoughts of 'Uhm, did you call the cops about this?' in my head while they are telling it... |
| 16:10.43 | Samot | I get curious, as I normally don't deal with this ITSPs end users... |
| 16:12.08 | Samot | "Like, what type of client is this." |
| 16:12.23 | Samot | And boom, Colon Hydrotherapy |
| 16:12.52 | Samot | I realized I'm hearing about patients who as she put "Given excellent service" being insane. |
| 16:13.15 | Samot | Because, like, they had an issue with their colon hydro |
| 16:13.23 | Samot | I'm not even sure how that works. |
| 16:13.45 | Samot | But the images of what it involves already make me scared. |
| 16:16.52 | Samot | They are in NY so I'm sure having your colon hosed out and in the most expensive and strange way is some how a social gold star. |
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| 16:29.30 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 16:35.16 | stefan27 | happy friday everyone |
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| 17:02.51 | qakhan | HI all, i am using ChanSpy(), is it possible to play a beep sound on spied channel before connect |
| 17:04.37 | [TK]D-Fender | What do the instructions say? |
| 17:04.47 | [TK]D-Fender | this isn't a guessing game |
| 17:04.52 | [TK]D-Fender | if it's an option then it's right there |
| 17:13.39 | qakhan | ok |
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| 17:59.39 | covale | hello |
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| 18:07.13 | covale | can anyone help me out with a Polycom bootrom and firmware zip files? I accidentally bricked my manager's phone =( |
| 18:07.56 | Chainsaw | covale: What model is the handset? |
| 18:08.05 | covale | or if someone can give me the actual file names or the current version number, I can just search. all I have been able to find online is bootrom_4_0_0_release_sig.zip |
| 18:08.27 | covale | soundpoint ip 501 SIP |
| 18:08.30 | Chainsaw | covale: It'd help to know what firmware it was on and what firmware you are trying to get to also. |
| 18:09.01 | covale | I think it has 4.0.0 Aug 2007 |
| 18:09.23 | covale | be nice to get the latest available for this model |
| 18:10.02 | covale | would that be 4.1, or is there newer? sorry, I am such a noob with these things. |
| 18:11.03 | Chainsaw | Let's have a look. I doubt 501 would handle UCS 4.x firmware. |
| 18:11.11 | Chainsaw | I think you should stick to 3.X SIP, and the corresponding bootloader. |
| 18:11.33 | covale | I will defer to your wisdom |
| 18:11.59 | Chainsaw | Just waiting for web pages to load. I'm on a train and the connectivity isn't fantastic. ETA 3 mins. |
| 18:12.02 | covale | would a pic of the about screen help? |
| 18:12.44 | covale | that's awesome! I really appreciate your help. |
| 18:13.12 | covale | and take all the time you need. I just need to get this thing unbricked before my manager gets back on Monday |
| 18:13.53 | Chainsaw | Let's go with the combined image: |
| 18:13.54 | Chainsaw | http://support.polycom.com/content/dam/polycom-support/products/Voice/soundpointip/downloads/spip_ssip_3_1_8_legacy_release_sig_combined/spip_ssip_3_1_8_legacy_release_sig_combined.zip |
| 18:14.13 | Chainsaw | That will work on the IP 501, even with the oldest boot ROMs. |
| 18:14.56 | Chainsaw | Backup your firmware directory before you start. |
| 18:16.50 | covale | https://snag.gy/syY6ri.jpg |
| 18:17.11 | covale | that's a pic of the about screen. does that mean I am on 4? |
| 18:17.24 | Chainsaw | That's your bootloader version, yes. |
| 18:17.38 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 18:18.05 | Chainsaw | But I'd still offer up the combined image I suggested. |
| 18:18.07 | Chainsaw | (Which is a single, giant sip.ld containing the SIP application for all different handsets) |
| 18:18.27 | covale | awesome, I will try that. thanks a bunch! |
| 18:18.50 | covale | do you recommend the ftp or tftp method of updating? |
| 18:18.58 | covale | or does it really matter? |
| 18:19.01 | Chainsaw | I use HTTP myself. |
| 18:19.19 | Chainsaw | TFTP is generally more reliable than FTP. |
| 18:19.29 | Chainsaw | (As the name suggests, Trivial FTP is a lot simpler) |
| 18:19.56 | covale | gotcha. this is my first time messing with one of these handsets. (as if it wasn't obvious) |
| 18:20.06 | Chainsaw | But FTP might have better logging and I suspect it will only upload logs from the handset to the server over FTP... so there's reasons for either. |
| 18:20.46 | Chainsaw | How are you provisioning? Are you setting a DHCP option with next-server for TFTP, or are you giving out a FTP/HTTP URL? |
| 18:22.15 | covale | was planning on setting up an ftp server and this phone is statically assigned. is that what you are asking? |
| 18:22.45 | *** join/#asterisk miralin (~Thunderbi@91.237.94.7) |
| 18:24.19 | Chainsaw | Well you'll have to modify your DHCP configuration to tell the phone where to fetch the firmware from. |
| 18:24.43 | Chainsaw | That's how I'd do it, anyway. |
| 18:26.19 | Chainsaw | You set a DHCP option that tells it where the firmware server is located. |
| 18:27.16 | *** join/#asterisk J0hnSteel (~J0hnSteel@92.55.116.178) |
| 18:27.23 | fauxalliance | option 55 iirc |
| 18:27.45 | fauxalliance | or maybe thats bootp |
| 18:28.08 | Chainsaw | It's either next-server for an IP or I think 61 for the string version. |
| 18:30.10 | Chainsaw | Fairly sure you're right with 55 for next-server, bootp & dhcp are rather compatible. |
| 18:31.06 | fauxalliance | i know |
| 18:31.09 | fauxalliance | thanks |
| 18:31.20 | fauxalliance | i checked the rfc ;) |
| 18:31.50 | Chainsaw | Well that's cheating. I didn't check my answers! |
| 18:32.38 | fauxalliance | can't be blamed for handing out bogus info, and I should stop second guessing myself anyway |
| 18:33.03 | fauxalliance | but a dhcp server with option 55 and a tftp server and you are done |
| 18:33.45 | Chainsaw | Agreed. Stick sip.ld in the root and it should recover the phone. |
| 18:33.51 | Chainsaw | Out of curiosity, how did you... break it? |
| 18:43.14 | *** join/#asterisk jkroon (~jkroon@165.16.204.162) |
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| 19:02.09 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 19:21.25 | covale | I'm sorry, were you asking me how I broke the phone, Chainsaw? |
| 19:25.11 | fauxalliance | yes |
| 19:25.16 | fauxalliance | i was curious too |
| 19:25.48 | covale | oh, from the reset menu I accidentally selected the "format option" |
| 19:25.54 | fauxalliance | cool |
| 19:26.20 | fauxalliance | not cool, but thanks for the answer |
| 19:26.37 | covale | sorry it took so long to reply, was setting up the ftp server and and a conf call while I was watching the phone connect and download the new files. |
| 19:26.48 | covale | really cool these little things are, even how old they are. |
| 19:28.03 | fauxalliance | polycoms, hell yes |
| 19:28.06 | fauxalliance | ?polycom |
| 19:28.21 | covale | yes, it's like ancient, but works like playschool |
| 19:28.42 | covale | soundpoint IP 501 |
| 19:29.17 | fauxalliance | ?mfpbx |
| 19:29.33 | fauxalliance | covale: they are great phones, rugged too |
| 19:29.46 | fauxalliance | i like cisco ones too, not the linksys ones. |
| 19:29.54 | covale | I think it's all better now. I was originally having an issue from the web interface on the "lines" screen where it would not save the settings or initiate the reboot once "sumbit" was clicked, so I was aiming to do a reset device and clicked on the format option. |
| 19:39.43 | *** join/#asterisk anpi (~crr@unaffiliated/anpi) |
| 19:43.33 | anpi | Hi, sipbroker used to permit US toll-free dialing without charge. Doesn't work today. Alternatives? |
| 20:05.55 | [TK]D-Fender | Alternative : Any other provider |
| 20:17.40 | Samot | anpi: What do you mean "without charge"? Are you referring to calling a toll-free destination or receiving calls to your toll-free number? |
| 20:18.02 | fauxalliance | termination, for free |
| 20:18.08 | Samot | Uhm. |
| 20:18.11 | fauxalliance | lol |
| 20:18.15 | Samot | That's the point of Toll-Free |
| 20:18.23 | Samot | So you terminate to it, without charge. |
| 20:18.26 | fauxalliance | yes |
| 20:18.52 | anpi | Samot: correct. Calling out to a US toll-free num while being outside of 'murica |
| 20:19.03 | Samot | OH |
| 20:19.06 | Samot | HAH. |
| 20:19.10 | fauxalliance | see also: lol |
| 20:19.15 | Samot | Yeah, that's not how Toll Free works. |
| 20:19.25 | Samot | They *may* permit it from Canada. |
| 20:19.30 | anpi | Samot: was working fine for years |
| 20:19.35 | fauxalliance | sipbroker _used_ to pass it free as a courtesy |
| 20:19.47 | Samot | ^^ That's what it sounds like. |
| 20:19.52 | [TK]D-Fender | So go find someone else who does. |
| 20:20.02 | fauxalliance | that was the question... full circle |
| 20:20.04 | Samot | I doubt you will. |
| 20:20.06 | anpi | that's what i asked.. |
| 20:20.08 | fauxalliance | me too |
| 20:20.31 | Samot | US Toll Free numbers are generally undialable outside of North America. |
| 20:20.50 | Samot | And even more, sometimes/most times not from Canada. |
| 20:20.57 | fauxalliance | https://www.voip-info.org/wiki/view/Toll+Free+Termination+Providers |
| 20:20.58 | anpi | i already found an alternative. gimme a sec to edit my * dialplan |
| 20:21.01 | fauxalliance | a whole wiki |
| 20:21.10 | Samot | Yeah from 2008 |
| 20:22.04 | fauxalliance | http://www.ideasip.com/ |
| 20:22.10 | Samot | voip-info.org should not be the source of anything you do, it's outdated and doesn't apply to most things. |
| 20:22.13 | fauxalliance | free, no registration, done in dialplan |
| 20:22.34 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 20:23.06 | fauxalliance | Samot: it's a list of providers, the coupon may or may not have expired |
| 20:23.10 | fauxalliance | s/coupon/link |
| 20:24.17 | anpi | got a peer dialout from a friend abroad. all good now. thanks |
| 20:25.37 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 20:39.36 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 21:22.38 | *** join/#asterisk Oatmeal (~Suzeanne@gateway/vpn/privateinternetaccess/suzeanne) |
| 21:27.16 | TheGallopingFox | when i listen to a voicemail, it cuts out before it has played |
| 21:27.19 | TheGallopingFox | the log shows this |
| 21:27.40 | TheGallopingFox | file.c: Unable to open digits/at (format (g722)): No such file or directory |
| 21:28.27 | TheGallopingFox | should asterisk transcode the voicemail .wav to g722? |
| 21:28.52 | TheGallopingFox | voicemail.conf has this: format=wav49 |
| 21:33.08 | Samot | That's looking in the digits folder |
| 21:33.13 | Samot | for at.g722 |
| 21:45.32 | TheGallopingFox | maybe asterisk doesnt transcode a voicemail wav |
| 21:45.45 | TheGallopingFox | so that means i need something like: format=wav49|gsm |
| 21:49.37 | Samot | What voicemail.wav? |
| 21:49.43 | Samot | What file are you talking about? |
| 21:50.46 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 22:09.09 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 22:13.24 | *** part/#asterisk kharwell (kharwell@nat/digium/x-fhjcclkifqpyoxnd) |
| 22:15.44 | TheGallopingFox | Samot: msg0000.WAV |
| 22:16.01 | TheGallopingFox | in /var/spool/asterisk/voicemail |
| 22:17.43 | TheGallopingFox | trouble is i cannot delete the messages now as voicemail stops before it plays |
| 22:18.27 | Samot | Sure you can. |
| 22:18.48 | Samot | rm -rf /var/spool/asterisk/voicemail/<account>/msg0000.* |
| 22:18.49 | Samot | Done. |
| 22:18.56 | Samot | Message gone. |
| 22:20.50 | TheGallopingFox | doesnt that effect a database somewhere? |
| 22:21.03 | Samot | What database? |
| 22:21.08 | TheGallopingFox | oh ok |
| 22:21.09 | TheGallopingFox | lol |
| 22:21.12 | TheGallopingFox | i will remove them manually |
| 22:21.13 | Samot | It's a flat file system by default. |
| 22:21.55 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:ba27:ebff:fee4:a52f) |
| 22:22.15 | TheGallopingFox | that worked |
| 22:22.29 | TheGallopingFox | i could of sworn my last asterisk setup managed to play voicemails |
| 22:22.34 | TheGallopingFox | through the phone |
| 22:22.53 | TheGallopingFox | i will test this now: format=wav49|gsm |
| 22:23.47 | TheGallopingFox | grrrr why does voip-info.org keep on giving me first results in google all the time |
| 22:24.10 | Samot | Because it's been around forever |
| 22:24.17 | Samot | People keep clicking through to it |
| 22:24.30 | Samot | asterisk github |
| 22:24.36 | Samot | All the sample files are there. |
| 22:25.00 | Samot | asterisk wiki <-- orders asterisk wiki results higher |
| 22:25.26 | Samot | Dial asterisk 14 wiki <-- Gets you more specific results |
| 22:25.48 | Samot | dial asterisk <-- Gets you viop-info and billy bob's VoIP blog from 2007 |
| 22:27.50 | fauxalliance | the monkeys! |
| 22:28.25 | TheGallopingFox | yeah i noticed lot of the stuff is old on void-info |
| 22:28.39 | TheGallopingFox | messed me up a few times following it |
| 22:29.28 | TheGallopingFox | right to clear things up, am i right in thinking that asterisk does _NOT_ transcode voicemail messages into suitable format so a phone can play a message? |
| 22:29.58 | TheGallopingFox | ie. if i use format=wav49, my polycom will not be able to playback the message |
| 22:30.07 | TheGallopingFox | however format=wav49|gsm will |
| 22:32.18 | Samot | or just wav |
| 22:32.43 | TheGallopingFox | or is it because i have this order |
| 22:32.44 | TheGallopingFox | disallow=all / allow=g722 / allow=alaw / allow=ulaw / allow=gsm |
| 22:32.57 | TheGallopingFox | and that explains this error message: |
| 22:33.11 | TheGallopingFox | file.c: Unable to open digits/at (format (g722)): No such file or directory |
| 22:33.33 | Samot | format=wav49|gsm|wav |
| 22:33.39 | Samot | use wav |
| 22:33.44 | TheGallopingFox | ok will test thanks |
| 22:33.57 | Samot | file.c: Unable to open digits/at (format (g722)): No such file or directory <-- This has nothing to do with the VM file |
| 22:34.04 | Samot | It's trying to read back the envelope |
| 22:34.13 | Samot | It's looking in the sounds directory DIGITS |
| 22:34.18 | Samot | for the file "at" |
| 22:34.34 | Samot | 5:33:07 PM <Samot> That's looking in the digits folder |
| 22:34.34 | Samot | 5:33:12 PM <Samot> for at.g722 |
| 22:34.42 | TheGallopingFox | oh somebody is ringing me :) |
| 22:35.43 | TheGallopingFox | ahhh it stopped as soon as it says 'first message recieved' |
| 22:36.06 | Samot | First message received "AT" <time date> |
| 22:36.12 | TheGallopingFox | [Sep 22 23:35:17] WARNING[24729][C-00000001] file.c: Unable to open digits/at (format (g722)): No such file or directory |
| 22:36.15 | TheGallopingFox | [Sep 22 23:35:17] WARNING[24729][C-00000001] say.c: Unable to play message digits/at |
| 22:36.19 | TheGallopingFox | oh i see |
| 22:36.30 | Samot | It can't find the files needed to playback the envelope details. |
| 22:36.32 | TheGallopingFox | im missing a sound file! |
| 22:36.40 | TheGallopingFox | ohhhhh |
| 22:36.40 | Samot | YES |
| 22:36.51 | TheGallopingFox | i mis-read the error |
| 22:36.55 | Samot | "No such file or directory" <-- Pretty clear |
| 22:37.05 | TheGallopingFox | right lets have a look |
| 22:38.50 | TheGallopingFox | at.alaw* / at.g723* / at.g729* / at.ulaw* / at.wav |
| 22:38.56 | TheGallopingFox | exist in digits |
| 22:39.04 | Samot | No g722 |
| 22:39.12 | Samot | Which is what it is looking for |
| 22:39.13 | TheGallopingFox | so why does the other sounds play |
| 22:40.36 | TheGallopingFox | so maybe i could put g729 at the begining of the codecs |
| 22:40.59 | TheGallopingFox | strange, nothing should play |
| 22:41.11 | TheGallopingFox | maybe its a corrupted file |
| 22:41.29 | TheGallopingFox | i liked her voice too :( |
| 22:48.34 | TheGallopingFox | ok i put the original voicemail sound files back, and it works |
| 22:49.40 | UncleKiwi | Hi there - just wondering if there is some good software that take the cdr from asterisk and then allow that data to be analysed ie for use in a callcenter so we can see how many calls per hour or day an agent was making or the average length of a call all that kind of thing |
| 22:50.37 | TheGallopingFox | i think my sounds are too old, updating them from here https://downloads.asterisk.org/pub/telephony/sounds/ |
| 22:52.01 | *** join/#asterisk iheartlinux (~jwpierce3@mail.trunkmasters.com) |
| 22:53.14 | TheGallopingFox | works now, no error messages!!!!!!! |
| 22:54.00 | TheGallopingFox | UncleKiwi: google is your friend |
| 22:54.19 | TheGallopingFox | unless you click on voip-info sites |
| 23:17.07 | UncleKiwi | yeah but i feel i have kfc fat in my brain |
| 23:26.44 | fauxalliance | kfc gravy. |
| 23:27.12 | fauxalliance | 'the dirty bird' we call it |
| 23:29.21 | UncleKiwi | ahaha |
| 23:35.10 | fauxalliance | there are call centre metrics available, but typically, in house , i'd cook the acd logs with a custom excel spreadsheet, AHT, THT, TCA, all that jazz |
| 23:35.21 | fauxalliance | then fire the 25th percentile |
| 23:35.46 | fauxalliance | untill they yelled about attirtion |
| 23:35.57 | fauxalliance | ahem attrition |
| 23:37.05 | fauxalliance | oh well |
